add dac continous latent space

dac-vae/audiotools/__init__.py

@@ -0,0 +1,10 @@
+__version__ = "0.7.4"
+from .core import AudioSignal
+from .core import STFTParams
+from .core import Meter
+from .core import util
+from . import metrics
+from . import data
+from . import ml
+from .data import datasets
+from .data import transforms

dac-vae/audiotools/core/__init__.py

@@ -0,0 +1,4 @@
+from . import util
+from .audio_signal import AudioSignal
+from .audio_signal import STFTParams
+from .loudness import Meter

dac-vae/audiotools/core/audio_signal.py

@@ -0,0 +1,1682 @@
+import copy
+import functools
+import hashlib
+import math
+import pathlib
+import tempfile
+import typing
+import warnings
+from collections import namedtuple
+from pathlib import Path
+
+import julius
+import numpy as np
+import soundfile
+import torch
+
+from . import util
+from .display import DisplayMixin
+from .dsp import DSPMixin
+from .effects import EffectMixin
+from .effects import ImpulseResponseMixin
+from .ffmpeg import FFMPEGMixin
+from .loudness import LoudnessMixin
+from .playback import PlayMixin
+from .whisper import WhisperMixin
+
+
+STFTParams = namedtuple(
+    "STFTParams",
+    ["window_length", "hop_length", "window_type", "match_stride", "padding_type"],
+)
+"""
+STFTParams object is a container that holds STFT parameters - window_length,
+hop_length, and window_type. Not all parameters need to be specified. Ones that
+are not specified will be inferred by the AudioSignal parameters.
+
+Parameters
+----------
+window_length : int, optional
+    Window length of STFT, by default ``0.032 * self.sample_rate``.
+hop_length : int, optional
+    Hop length of STFT, by default ``window_length // 4``.
+window_type : str, optional
+    Type of window to use, by default ``sqrt\_hann``.
+match_stride : bool, optional
+    Whether to match the stride of convolutional layers, by default False
+padding_type : str, optional
+    Type of padding to use, by default 'reflect'
+"""
+STFTParams.__new__.__defaults__ = (None, None, None, None, None)
+
+
+class AudioSignal(
+    EffectMixin,
+    LoudnessMixin,
+    PlayMixin,
+    ImpulseResponseMixin,
+    DSPMixin,
+    DisplayMixin,
+    FFMPEGMixin,
+    WhisperMixin,
+):
+    """This is the core object of this library. Audio is always
+    loaded into an AudioSignal, which then enables all the features
+    of this library, including audio augmentations, I/O, playback,
+    and more.
+
+    The structure of this object is that the base functionality
+    is defined in ``core/audio_signal.py``, while extensions to
+    that functionality are defined in the other ``core/*.py``
+    files. For example, all the display-based functionality
+    (e.g. plot spectrograms, waveforms, write to tensorboard)
+    are in ``core/display.py``.
+
+    Parameters
+    ----------
+    audio_path_or_array : typing.Union[torch.Tensor, str, Path, np.ndarray]
+        Object to create AudioSignal from. Can be a tensor, numpy array,
+        or a path to a file. The file is always reshaped to
+    sample_rate : int, optional
+        Sample rate of the audio. If different from underlying file, resampling is
+        performed. If passing in an array or tensor, this must be defined,
+        by default None
+    stft_params : STFTParams, optional
+        Parameters of STFT to use. , by default None
+    offset : float, optional
+        Offset in seconds to read from file, by default 0
+    duration : float, optional
+        Duration in seconds to read from file, by default None
+    device : str, optional
+        Device to load audio onto, by default None
+
+    Examples
+    --------
+    Loading an AudioSignal from an array, at a sample rate of
+    44100.
+
+    >>> signal = AudioSignal(torch.randn(5*44100), 44100)
+
+    Note, the signal is reshaped to have a batch size, and one
+    audio channel:
+
+    >>> print(signal.shape)
+    (1, 1, 44100)
+
+    You can treat AudioSignals like tensors, and many of the same
+    functions you might use on tensors are defined for AudioSignals
+    as well:
+
+    >>> signal.to("cuda")
+    >>> signal.cuda()
+    >>> signal.clone()
+    >>> signal.detach()
+
+    Indexing AudioSignals returns an AudioSignal:
+
+    >>> signal[..., 3*44100:4*44100]
+
+    The above signal is 1 second long, and is also an AudioSignal.
+    """
+
+    def __init__(
+        self,
+        audio_path_or_array: typing.Union[torch.Tensor, str, Path, np.ndarray],
+        sample_rate: int = None,
+        stft_params: STFTParams = None,
+        offset: float = 0,
+        duration: float = None,
+        device: str = None,
+    ):
+        audio_path = None
+        audio_array = None
+
+        if isinstance(audio_path_or_array, str):
+            audio_path = audio_path_or_array
+        elif isinstance(audio_path_or_array, pathlib.Path):
+            audio_path = audio_path_or_array
+        elif isinstance(audio_path_or_array, np.ndarray):
+            audio_array = audio_path_or_array
+        elif torch.is_tensor(audio_path_or_array):
+            audio_array = audio_path_or_array
+        else:
+            raise ValueError(
+                "audio_path_or_array must be either a Path, "
+                "string, numpy array, or torch Tensor!"
+            )
+
+        self.path_to_file = None
+
+        self.audio_data = None
+        self.sources = None  # List of AudioSignal objects.
+        self.stft_data = None
+        if audio_path is not None:
+            self.load_from_file(
+                audio_path, offset=offset, duration=duration, device=device
+            )
+        elif audio_array is not None:
+            assert sample_rate is not None, "Must set sample rate!"
+            self.load_from_array(audio_array, sample_rate, device=device)
+
+        self.window = None
+        self.stft_params = stft_params
+
+        self.metadata = {
+            "offset": offset,
+            "duration": duration,
+        }
+
+    @property
+    def path_to_input_file(
+        self,
+    ):
+        """
+        Path to input file, if it exists.
+        Alias to ``path_to_file`` for backwards compatibility
+        """
+        return self.path_to_file
+
+    @classmethod
+    def excerpt(
+        cls,
+        audio_path: typing.Union[str, Path],
+        offset: float = None,
+        duration: float = None,
+        state: typing.Union[np.random.RandomState, int] = None,
+        **kwargs,
+    ):
+        """Randomly draw an excerpt of ``duration`` seconds from an
+        audio file specified at ``audio_path``, between ``offset`` seconds
+        and end of file. ``state`` can be used to seed the random draw.
+
+        Parameters
+        ----------
+        audio_path : typing.Union[str, Path]
+            Path to audio file to grab excerpt from.
+        offset : float, optional
+            Lower bound for the start time, in seconds drawn from
+            the file, by default None.
+        duration : float, optional
+            Duration of excerpt, in seconds, by default None
+        state : typing.Union[np.random.RandomState, int], optional
+            RandomState or seed of random state, by default None
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal containing excerpt.
+
+        Examples
+        --------
+        >>> signal = AudioSignal.excerpt("path/to/audio", duration=5)
+        """
+        info = util.info(audio_path)
+        total_duration = info.duration
+
+        state = util.random_state(state)
+        lower_bound = 0 if offset is None else offset
+        upper_bound = max(total_duration - duration, 0)
+        offset = state.uniform(lower_bound, upper_bound)
+
+        signal = cls(audio_path, offset=offset, duration=duration, **kwargs)
+        signal.metadata["offset"] = offset
+        signal.metadata["duration"] = duration
+
+        return signal
+
+    @classmethod
+    def salient_excerpt(
+        cls,
+        audio_path: typing.Union[str, Path],
+        loudness_cutoff: float = None,
+        num_tries: int = 8,
+        state: typing.Union[np.random.RandomState, int] = None,
+        **kwargs,
+    ):
+        """Similar to AudioSignal.excerpt, except it extracts excerpts only
+        if they are above a specified loudness threshold, which is computed via
+        a fast LUFS routine.
+
+        Parameters
+        ----------
+        audio_path : typing.Union[str, Path]
+            Path to audio file to grab excerpt from.
+        loudness_cutoff : float, optional
+            Loudness threshold in dB. Typical values are ``-40, -60``,
+            etc, by default None
+        num_tries : int, optional
+            Number of tries to grab an excerpt above the threshold
+            before giving up, by default 8.
+        state : typing.Union[np.random.RandomState, int], optional
+            RandomState or seed of random state, by default None
+        kwargs : dict
+            Keyword arguments to AudioSignal.excerpt
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal containing excerpt.
+
+
+        .. warning::
+            if ``num_tries`` is set to None, ``salient_excerpt`` may try forever, which can
+            result in an infinite loop if ``audio_path`` does not have
+            any loud enough excerpts.
+
+        Examples
+        --------
+        >>> signal = AudioSignal.salient_excerpt(
+                "path/to/audio",
+                loudness_cutoff=-40,
+                duration=5
+            )
+        """
+        state = util.random_state(state)
+        if loudness_cutoff is None:
+            excerpt = cls.excerpt(audio_path, state=state, **kwargs)
+        else:
+            loudness = -np.inf
+            num_try = 0
+            while loudness <= loudness_cutoff:
+                excerpt = cls.excerpt(audio_path, state=state, **kwargs)
+                loudness = excerpt.loudness()
+                num_try += 1
+                if num_tries is not None and num_try >= num_tries:
+                    break
+        return excerpt
+
+    @classmethod
+    def zeros(
+        cls,
+        duration: float,
+        sample_rate: int,
+        num_channels: int = 1,
+        batch_size: int = 1,
+        **kwargs,
+    ):
+        """Helper function create an AudioSignal of all zeros.
+
+        Parameters
+        ----------
+        duration : float
+            Duration of AudioSignal
+        sample_rate : int
+            Sample rate of AudioSignal
+        num_channels : int, optional
+            Number of channels, by default 1
+        batch_size : int, optional
+            Batch size, by default 1
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal containing all zeros.
+
+        Examples
+        --------
+        Generate 5 seconds of all zeros at a sample rate of 44100.
+
+        >>> signal = AudioSignal.zeros(5.0, 44100)
+        """
+        n_samples = int(duration * sample_rate)
+        return cls(
+            torch.zeros(batch_size, num_channels, n_samples), sample_rate, **kwargs
+        )
+
+    @classmethod
+    def wave(
+        cls,
+        frequency: float,
+        duration: float,
+        sample_rate: int,
+        num_channels: int = 1,
+        shape: str = "sine",
+        **kwargs,
+    ):
+        """
+        Generate a waveform of a given frequency and shape.
+
+        Parameters
+        ----------
+        frequency : float
+            Frequency of the waveform
+        duration : float
+            Duration of the waveform
+        sample_rate : int
+            Sample rate of the waveform
+        num_channels : int, optional
+            Number of channels, by default 1
+        shape : str, optional
+            Shape of the waveform, by default "saw"
+            One of "sawtooth", "square", "sine", "triangle"
+        kwargs : dict
+            Keyword arguments to AudioSignal
+        """
+        n_samples = int(duration * sample_rate)
+        t = torch.linspace(0, duration, n_samples)
+        if shape == "sawtooth":
+            from scipy.signal import sawtooth
+
+            wave_data = sawtooth(2 * np.pi * frequency * t, 0.5)
+        elif shape == "square":
+            from scipy.signal import square
+
+            wave_data = square(2 * np.pi * frequency * t)
+        elif shape == "sine":
+            wave_data = np.sin(2 * np.pi * frequency * t)
+        elif shape == "triangle":
+            from scipy.signal import sawtooth
+
+            # frequency is doubled by the abs call, so omit the 2 in 2pi
+            wave_data = sawtooth(np.pi * frequency * t, 0.5)
+            wave_data = -np.abs(wave_data) * 2 + 1
+        else:
+            raise ValueError(f"Invalid shape {shape}")
+
+        wave_data = torch.tensor(wave_data, dtype=torch.float32)
+        wave_data = wave_data.unsqueeze(0).unsqueeze(0).repeat(1, num_channels, 1)
+        return cls(wave_data, sample_rate, **kwargs)
+
+    @classmethod
+    def batch(
+        cls,
+        audio_signals: list,
+        pad_signals: bool = False,
+        truncate_signals: bool = False,
+        resample: bool = False,
+        dim: int = 0,
+    ):
+        """Creates a batched AudioSignal from a list of AudioSignals.
+
+        Parameters
+        ----------
+        audio_signals : list[AudioSignal]
+            List of AudioSignal objects
+        pad_signals : bool, optional
+            Whether to pad signals to length of the maximum length
+            AudioSignal in the list, by default False
+        truncate_signals : bool, optional
+            Whether to truncate signals to length of shortest length
+            AudioSignal in the list, by default False
+        resample : bool, optional
+            Whether to resample AudioSignal to the sample rate of
+            the first AudioSignal in the list, by default False
+        dim : int, optional
+            Dimension along which to batch the signals.
+
+        Returns
+        -------
+        AudioSignal
+            Batched AudioSignal.
+
+        Raises
+        ------
+        RuntimeError
+            If not all AudioSignals are the same sample rate, and
+            ``resample=False``, an error is raised.
+        RuntimeError
+            If not all AudioSignals are the same the length, and
+            both ``pad_signals=False`` and ``truncate_signals=False``,
+            an error is raised.
+
+        Examples
+        --------
+        Batching a bunch of random signals:
+
+        >>> signal_list = [AudioSignal(torch.randn(44100), 44100) for _ in range(10)]
+        >>> signal = AudioSignal.batch(signal_list)
+        >>> print(signal.shape)
+        (10, 1, 44100)
+
+        """
+        signal_lengths = [x.signal_length for x in audio_signals]
+        sample_rates = [x.sample_rate for x in audio_signals]
+
+        if len(set(sample_rates)) != 1:
+            if resample:
+                for x in audio_signals:
+                    x.resample(sample_rates[0])
+            else:
+                raise RuntimeError(
+                    f"Not all signals had the same sample rate! Got {sample_rates}. "
+                    f"All signals must have the same sample rate, or resample must be True. "
+                )
+
+        if len(set(signal_lengths)) != 1:
+            if pad_signals:
+                max_length = max(signal_lengths)
+                for x in audio_signals:
+                    pad_len = max_length - x.signal_length
+                    x.zero_pad(0, pad_len)
+            elif truncate_signals:
+                min_length = min(signal_lengths)
+                for x in audio_signals:
+                    x.truncate_samples(min_length)
+            else:
+                raise RuntimeError(
+                    f"Not all signals had the same length! Got {signal_lengths}. "
+                    f"All signals must be the same length, or pad_signals/truncate_signals "
+                    f"must be True. "
+                )
+        # Concatenate along the specified dimension (default 0)
+        audio_data = torch.cat([x.audio_data for x in audio_signals], dim=dim)
+        audio_paths = [x.path_to_file for x in audio_signals]
+
+        batched_signal = cls(
+            audio_data,
+            sample_rate=audio_signals[0].sample_rate,
+        )
+        batched_signal.path_to_file = audio_paths
+        return batched_signal
+
+    # I/O
+    def load_from_file(
+        self,
+        audio_path: typing.Union[str, Path],
+        offset: float,
+        duration: float,
+        device: str = "cpu",
+    ):
+        """Loads data from file. Used internally when AudioSignal
+        is instantiated with a path to a file.
+
+        Parameters
+        ----------
+        audio_path : typing.Union[str, Path]
+            Path to file
+        offset : float
+            Offset in seconds
+        duration : float
+            Duration in seconds
+        device : str, optional
+            Device to put AudioSignal on, by default "cpu"
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal loaded from file
+        """
+        import librosa
+
+        data, sample_rate = librosa.load(
+            audio_path,
+            offset=offset,
+            duration=duration,
+            sr=None,
+            mono=False,
+        )
+        data = util.ensure_tensor(data)
+        if data.shape[-1] == 0:
+            raise RuntimeError(
+                f"Audio file {audio_path} with offset {offset} and duration {duration} is empty!"
+            )
+
+        if data.ndim < 2:
+            data = data.unsqueeze(0)
+        if data.ndim < 3:
+            data = data.unsqueeze(0)
+        self.audio_data = data
+
+        self.original_signal_length = self.signal_length
+
+        self.sample_rate = sample_rate
+        self.path_to_file = audio_path
+        return self.to(device)
+
+    def load_from_array(
+        self,
+        audio_array: typing.Union[torch.Tensor, np.ndarray],
+        sample_rate: int,
+        device: str = "cpu",
+    ):
+        """Loads data from array, reshaping it to be exactly 3
+        dimensions. Used internally when AudioSignal is called
+        with a tensor or an array.
+
+        Parameters
+        ----------
+        audio_array : typing.Union[torch.Tensor, np.ndarray]
+            Array/tensor of audio of samples.
+        sample_rate : int
+            Sample rate of audio
+        device : str, optional
+            Device to move audio onto, by default "cpu"
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal loaded from array
+        """
+        audio_data = util.ensure_tensor(audio_array)
+
+        if audio_data.dtype == torch.double:
+            audio_data = audio_data.float()
+
+        if audio_data.ndim < 2:
+            audio_data = audio_data.unsqueeze(0)
+        if audio_data.ndim < 3:
+            audio_data = audio_data.unsqueeze(0)
+        self.audio_data = audio_data
+
+        self.original_signal_length = self.signal_length
+
+        self.sample_rate = sample_rate
+        return self.to(device)
+
+    def write(self, audio_path: typing.Union[str, Path]):
+        """Writes audio to a file. Only writes the audio
+        that is in the very first item of the batch. To write other items
+        in the batch, index the signal along the batch dimension
+        before writing. After writing, the signal's ``path_to_file``
+        attribute is updated to the new path.
+
+        Parameters
+        ----------
+        audio_path : typing.Union[str, Path]
+            Path to write audio to.
+
+        Returns
+        -------
+        AudioSignal
+            Returns original AudioSignal, so you can use this in a fluent
+            interface.
+
+        Examples
+        --------
+        Creating and writing a signal to disk:
+
+        >>> signal = AudioSignal(torch.randn(10, 1, 44100), 44100)
+        >>> signal.write("/tmp/out.wav")
+
+        Writing a different element of the batch:
+
+        >>> signal[5].write("/tmp/out.wav")
+
+        Using this in a fluent interface:
+
+        >>> signal.write("/tmp/original.wav").low_pass(4000).write("/tmp/lowpass.wav")
+
+        """
+        if self.audio_data[0].abs().max() > 1:
+            warnings.warn("Audio amplitude > 1 clipped when saving")
+        soundfile.write(str(audio_path), self.audio_data[0].numpy().T, self.sample_rate)
+
+        self.path_to_file = audio_path
+        return self
+
+    def deepcopy(self):
+        """Copies the signal and all of its attributes.
+
+        Returns
+        -------
+        AudioSignal
+            Deep copy of the audio signal.
+        """
+        return copy.deepcopy(self)
+
+    def copy(self):
+        """Shallow copy of signal.
+
+        Returns
+        -------
+        AudioSignal
+            Shallow copy of the audio signal.
+        """
+        return copy.copy(self)
+
+    def clone(self):
+        """Clones all tensors contained in the AudioSignal,
+        and returns a copy of the signal with everything
+        cloned. Useful when using AudioSignal within autograd
+        computation graphs.
+
+        Relevant attributes are the stft data, the audio data,
+        and the loudness of the file.
+
+        Returns
+        -------
+        AudioSignal
+            Clone of AudioSignal.
+        """
+        clone = type(self)(
+            self.audio_data.clone(),
+            self.sample_rate,
+            stft_params=self.stft_params,
+        )
+        if self.stft_data is not None:
+            clone.stft_data = self.stft_data.clone()
+        if self._loudness is not None:
+            clone._loudness = self._loudness.clone()
+        clone.path_to_file = copy.deepcopy(self.path_to_file)
+        clone.metadata = copy.deepcopy(self.metadata)
+        return clone
+
+    def detach(self):
+        """Detaches tensors contained in AudioSignal.
+
+        Relevant attributes are the stft data, the audio data,
+        and the loudness of the file.
+
+        Returns
+        -------
+        AudioSignal
+            Same signal, but with all tensors detached.
+        """
+        if self._loudness is not None:
+            self._loudness = self._loudness.detach()
+        if self.stft_data is not None:
+            self.stft_data = self.stft_data.detach()
+
+        self.audio_data = self.audio_data.detach()
+        return self
+
+    def hash(self):
+        """Writes the audio data to a temporary file, and then
+        hashes it using hashlib. Useful for creating a file
+        name based on the audio content.
+
+        Returns
+        -------
+        str
+            Hash of audio data.
+
+        Examples
+        --------
+        Creating a signal, and writing it to a unique file name:
+
+        >>> signal = AudioSignal(torch.randn(44100), 44100)
+        >>> hash = signal.hash()
+        >>> signal.write(f"{hash}.wav")
+
+        """
+        with tempfile.NamedTemporaryFile(suffix=".wav") as f:
+            self.write(f.name)
+            h = hashlib.sha256()
+            b = bytearray(128 * 1024)
+            mv = memoryview(b)
+            with open(f.name, "rb", buffering=0) as f:
+                for n in iter(lambda: f.readinto(mv), 0):
+                    h.update(mv[:n])
+            file_hash = h.hexdigest()
+        return file_hash
+
+    # Signal operations
+    def to_mono(self):
+        """Converts audio data to mono audio, by taking the mean
+        along the channels dimension.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with mean of channels.
+        """
+        self.audio_data = self.audio_data.mean(1, keepdim=True)
+        return self
+
+    def resample(self, sample_rate: int):
+        """Resamples the audio, using sinc interpolation. This works on both
+        cpu and gpu, and is much faster on gpu.
+
+        Parameters
+        ----------
+        sample_rate : int
+            Sample rate to resample to.
+
+        Returns
+        -------
+        AudioSignal
+            Resampled AudioSignal
+        """
+        if sample_rate == self.sample_rate:
+            return self
+        self.audio_data = julius.resample_frac(
+            self.audio_data, self.sample_rate, sample_rate
+        )
+        self.sample_rate = sample_rate
+        return self
+
+    # Tensor operations
+    def to(self, device: str):
+        """Moves all tensors contained in signal to the specified device.
+
+        Parameters
+        ----------
+        device : str
+            Device to move AudioSignal onto. Typical values are
+            "cuda", "cpu", or "cuda:n" to specify the nth gpu.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with all tensors moved to specified device.
+        """
+        if self._loudness is not None:
+            self._loudness = self._loudness.to(device)
+        if self.stft_data is not None:
+            self.stft_data = self.stft_data.to(device)
+        if self.audio_data is not None:
+            self.audio_data = self.audio_data.to(device)
+        return self
+
+    def float(self):
+        """Calls ``.float()`` on ``self.audio_data``.
+
+        Returns
+        -------
+        AudioSignal
+        """
+        self.audio_data = self.audio_data.float()
+        return self
+
+    def cpu(self):
+        """Moves AudioSignal to cpu.
+
+        Returns
+        -------
+        AudioSignal
+        """
+        return self.to("cpu")
+
+    def cuda(self):  # pragma: no cover
+        """Moves AudioSignal to cuda.
+
+        Returns
+        -------
+        AudioSignal
+        """
+        return self.to("cuda")
+
+    def numpy(self):
+        """Detaches ``self.audio_data``, moves to cpu, and converts to numpy.
+
+        Returns
+        -------
+        np.ndarray
+            Audio data as a numpy array.
+        """
+        return self.audio_data.detach().cpu().numpy()
+
+    def zero_pad(self, before: int, after: int):
+        """Zero pads the audio_data tensor before and after.
+
+        Parameters
+        ----------
+        before : int
+            How many zeros to prepend to audio.
+        after : int
+            How many zeros to append to audio.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with padding applied.
+        """
+        self.audio_data = torch.nn.functional.pad(self.audio_data, (before, after))
+        return self
+
+    def zero_pad_to(self, length: int, mode: str = "after"):
+        """Pad with zeros to a specified length, either before or after
+        the audio data.
+
+        Parameters
+        ----------
+        length : int
+            Length to pad to
+        mode : str, optional
+            Whether to prepend or append zeros to signal, by default "after"
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with padding applied.
+        """
+        if mode == "before":
+            self.zero_pad(max(length - self.signal_length, 0), 0)
+        elif mode == "after":
+            self.zero_pad(0, max(length - self.signal_length, 0))
+        return self
+
+    def trim(self, before: int, after: int):
+        """Trims the audio_data tensor before and after.
+
+        Parameters
+        ----------
+        before : int
+            How many samples to trim from beginning.
+        after : int
+            How many samples to trim from end.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with trimming applied.
+        """
+        if after == 0:
+            self.audio_data = self.audio_data[..., before:]
+        else:
+            self.audio_data = self.audio_data[..., before:-after]
+        return self
+
+    def truncate_samples(self, length_in_samples: int):
+        """Truncate signal to specified length.
+
+        Parameters
+        ----------
+        length_in_samples : int
+            Truncate to this many samples.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with truncation applied.
+        """
+        self.audio_data = self.audio_data[..., :length_in_samples]
+        return self
+
+    @property
+    def device(self):
+        """Get device that AudioSignal is on.
+
+        Returns
+        -------
+        torch.device
+            Device that AudioSignal is on.
+        """
+        if self.audio_data is not None:
+            device = self.audio_data.device
+        elif self.stft_data is not None:
+            device = self.stft_data.device
+        return device
+
+    # Properties
+    @property
+    def audio_data(self):
+        """Returns the audio data tensor in the object.
+
+        Audio data is always of the shape
+        (batch_size, num_channels, num_samples). If value has less
+        than 3 dims (e.g. is (num_channels, num_samples)), then it will
+        be reshaped to (1, num_channels, num_samples) - a batch size of 1.
+
+        Parameters
+        ----------
+        data : typing.Union[torch.Tensor, np.ndarray]
+            Audio data to set.
+
+        Returns
+        -------
+        torch.Tensor
+            Audio samples.
+        """
+        return self._audio_data
+
+    @audio_data.setter
+    def audio_data(self, data: typing.Union[torch.Tensor, np.ndarray]):
+        if data is not None:
+            assert torch.is_tensor(data), "audio_data should be torch.Tensor"
+            assert data.ndim == 3, "audio_data should be 3-dim (B, C, T)"
+        self._audio_data = data
+        # Old loudness value not guaranteed to be right, reset it.
+        self._loudness = None
+        return
+
+    # alias for audio_data
+    samples = audio_data
+
+    @property
+    def stft_data(self):
+        """Returns the STFT data inside the signal. Shape is
+        (batch, channels, frequencies, time).
+
+        Returns
+        -------
+        torch.Tensor
+            Complex spectrogram data.
+        """
+        return self._stft_data
+
+    @stft_data.setter
+    def stft_data(self, data: typing.Union[torch.Tensor, np.ndarray]):
+        if data is not None:
+            assert torch.is_tensor(data) and torch.is_complex(data)
+            if self.stft_data is not None and self.stft_data.shape != data.shape:
+                warnings.warn("stft_data changed shape")
+        self._stft_data = data
+        return
+
+    @property
+    def batch_size(self):
+        """Batch size of audio signal.
+
+        Returns
+        -------
+        int
+            Batch size of signal.
+        """
+        return self.audio_data.shape[0]
+
+    @property
+    def signal_length(self):
+        """Length of audio signal.
+
+        Returns
+        -------
+        int
+            Length of signal in samples.
+        """
+        return self.audio_data.shape[-1]
+
+    # alias for signal_length
+    length = signal_length
+
+    @property
+    def shape(self):
+        """Shape of audio data.
+
+        Returns
+        -------
+        tuple
+            Shape of audio data.
+        """
+        return self.audio_data.shape
+
+    @property
+    def signal_duration(self):
+        """Length of audio signal in seconds.
+
+        Returns
+        -------
+        float
+            Length of signal in seconds.
+        """
+        return self.signal_length / self.sample_rate
+
+    # alias for signal_duration
+    duration = signal_duration
+
+    @property
+    def num_channels(self):
+        """Number of audio channels.
+
+        Returns
+        -------
+        int
+            Number of audio channels.
+        """
+        return self.audio_data.shape[1]
+
+    # STFT
+    @staticmethod
+    @functools.lru_cache(None)
+    def get_window(window_type: str, window_length: int, device: str):
+        """Wrapper around scipy.signal.get_window so one can also get the
+        popular sqrt-hann window. This function caches for efficiency
+        using functools.lru\_cache.
+
+        Parameters
+        ----------
+        window_type : str
+            Type of window to get
+        window_length : int
+            Length of the window
+        device : str
+            Device to put window onto.
+
+        Returns
+        -------
+        torch.Tensor
+            Window returned by scipy.signal.get_window, as a tensor.
+        """
+        from scipy import signal
+
+        if window_type == "average":
+            window = np.ones(window_length) / window_length
+        elif window_type == "sqrt_hann":
+            window = np.sqrt(signal.get_window("hann", window_length))
+        else:
+            window = signal.get_window(window_type, window_length)
+        window = torch.from_numpy(window).to(device).float()
+        return window
+
+    @property
+    def stft_params(self):
+        """Returns STFTParams object, which can be re-used to other
+        AudioSignals.
+
+        This property can be set as well. If values are not defined in STFTParams,
+        they are inferred automatically from the signal properties. The default is to use
+        32ms windows, with 8ms hop length, and the square root of the hann window.
+
+        Returns
+        -------
+        STFTParams
+            STFT parameters for the AudioSignal.
+
+        Examples
+        --------
+        >>> stft_params = STFTParams(128, 32)
+        >>> signal1 = AudioSignal(torch.randn(44100), 44100, stft_params=stft_params)
+        >>> signal2 = AudioSignal(torch.randn(44100), 44100, stft_params=signal1.stft_params)
+        >>> signal1.stft_params = STFTParams() # Defaults
+        """
+        return self._stft_params
+
+    @stft_params.setter
+    def stft_params(self, value: STFTParams):
+        default_win_len = int(2 ** (np.ceil(np.log2(0.032 * self.sample_rate))))
+        default_hop_len = default_win_len // 4
+        default_win_type = "hann"
+        default_match_stride = False
+        default_padding_type = "reflect"
+
+        default_stft_params = STFTParams(
+            window_length=default_win_len,
+            hop_length=default_hop_len,
+            window_type=default_win_type,
+            match_stride=default_match_stride,
+            padding_type=default_padding_type,
+        )._asdict()
+
+        value = value._asdict() if value else default_stft_params
+
+        for key in default_stft_params:
+            if value[key] is None:
+                value[key] = default_stft_params[key]
+
+        self._stft_params = STFTParams(**value)
+        self.stft_data = None
+
+    def compute_stft_padding(
+        self, window_length: int, hop_length: int, match_stride: bool
+    ):
+        """Compute how the STFT should be padded, based on match\_stride.
+
+        Parameters
+        ----------
+        window_length : int
+            Window length of STFT.
+        hop_length : int
+            Hop length of STFT.
+        match_stride : bool
+            Whether or not to match stride, making the STFT have the same alignment as
+            convolutional layers.
+
+        Returns
+        -------
+        tuple
+            Amount to pad on either side of audio.
+        """
+        length = self.signal_length
+
+        if match_stride:
+            assert (
+                hop_length == window_length // 4
+            ), "For match_stride, hop must equal n_fft // 4"
+            right_pad = math.ceil(length / hop_length) * hop_length - length
+            pad = (window_length - hop_length) // 2
+        else:
+            right_pad = 0
+            pad = 0
+
+        return right_pad, pad
+
+    def stft(
+        self,
+        window_length: int = None,
+        hop_length: int = None,
+        window_type: str = None,
+        match_stride: bool = None,
+        padding_type: str = None,
+    ):
+        """Computes the short-time Fourier transform of the audio data,
+        with specified STFT parameters.
+
+        Parameters
+        ----------
+        window_length : int, optional
+            Window length of STFT, by default ``0.032 * self.sample_rate``.
+        hop_length : int, optional
+            Hop length of STFT, by default ``window_length // 4``.
+        window_type : str, optional
+            Type of window to use, by default ``sqrt\_hann``.
+        match_stride : bool, optional
+            Whether to match the stride of convolutional layers, by default False
+        padding_type : str, optional
+            Type of padding to use, by default 'reflect'
+
+        Returns
+        -------
+        torch.Tensor
+            STFT of audio data.
+
+        Examples
+        --------
+        Compute the STFT of an AudioSignal:
+
+        >>> signal = AudioSignal(torch.randn(44100), 44100)
+        >>> signal.stft()
+
+        Vary the window and hop length:
+
+        >>> stft_params = [STFTParams(128, 32), STFTParams(512, 128)]
+        >>> for stft_param in stft_params:
+        >>>     signal.stft_params = stft_params
+        >>>     signal.stft()
+
+        """
+        window_length = (
+            self.stft_params.window_length
+            if window_length is None
+            else int(window_length)
+        )
+        hop_length = (
+            self.stft_params.hop_length if hop_length is None else int(hop_length)
+        )
+        window_type = (
+            self.stft_params.window_type if window_type is None else window_type
+        )
+        match_stride = (
+            self.stft_params.match_stride if match_stride is None else match_stride
+        )
+        padding_type = (
+            self.stft_params.padding_type if padding_type is None else padding_type
+        )
+
+        window = self.get_window(window_type, window_length, self.audio_data.device)
+        window = window.to(self.audio_data.device)
+
+        audio_data = self.audio_data
+        right_pad, pad = self.compute_stft_padding(
+            window_length, hop_length, match_stride
+        )
+        audio_data = torch.nn.functional.pad(
+            audio_data, (pad, pad + right_pad), padding_type
+        )
+        stft_data = torch.stft(
+            audio_data.reshape(-1, audio_data.shape[-1]),
+            n_fft=window_length,
+            hop_length=hop_length,
+            window=window,
+            return_complex=True,
+            center=True,
+        )
+        _, nf, nt = stft_data.shape
+        stft_data = stft_data.reshape(self.batch_size, self.num_channels, nf, nt)
+
+        if match_stride:
+            # Drop first two and last two frames, which are added
+            # because of padding. Now num_frames * hop_length = num_samples.
+            stft_data = stft_data[..., 2:-2]
+        self.stft_data = stft_data
+
+        return stft_data
+
+    def istft(
+        self,
+        window_length: int = None,
+        hop_length: int = None,
+        window_type: str = None,
+        match_stride: bool = None,
+        length: int = None,
+    ):
+        """Computes inverse STFT and sets it to audio\_data.
+
+        Parameters
+        ----------
+        window_length : int, optional
+            Window length of STFT, by default ``0.032 * self.sample_rate``.
+        hop_length : int, optional
+            Hop length of STFT, by default ``window_length // 4``.
+        window_type : str, optional
+            Type of window to use, by default ``sqrt\_hann``.
+        match_stride : bool, optional
+            Whether to match the stride of convolutional layers, by default False
+        length : int, optional
+            Original length of signal, by default None
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with istft applied.
+
+        Raises
+        ------
+        RuntimeError
+            Raises an error if stft was not called prior to istft on the signal,
+            or if stft_data is not set.
+        """
+        if self.stft_data is None:
+            raise RuntimeError("Cannot do inverse STFT without self.stft_data!")
+
+        window_length = (
+            self.stft_params.window_length
+            if window_length is None
+            else int(window_length)
+        )
+        hop_length = (
+            self.stft_params.hop_length if hop_length is None else int(hop_length)
+        )
+        window_type = (
+            self.stft_params.window_type if window_type is None else window_type
+        )
+        match_stride = (
+            self.stft_params.match_stride if match_stride is None else match_stride
+        )
+
+        window = self.get_window(window_type, window_length, self.stft_data.device)
+
+        nb, nch, nf, nt = self.stft_data.shape
+        stft_data = self.stft_data.reshape(nb * nch, nf, nt)
+        right_pad, pad = self.compute_stft_padding(
+            window_length, hop_length, match_stride
+        )
+
+        if length is None:
+            length = self.original_signal_length
+            length = length + 2 * pad + right_pad
+
+        if match_stride:
+            # Zero-pad the STFT on either side, putting back the frames that were
+            # dropped in stft().
+            stft_data = torch.nn.functional.pad(stft_data, (2, 2))
+
+        audio_data = torch.istft(
+            stft_data,
+            n_fft=window_length,
+            hop_length=hop_length,
+            window=window,
+            length=length,
+            center=True,
+        )
+        audio_data = audio_data.reshape(nb, nch, -1)
+        if match_stride:
+            audio_data = audio_data[..., pad : -(pad + right_pad)]
+        self.audio_data = audio_data
+
+        return self
+
+    @staticmethod
+    @functools.lru_cache(None)
+    def get_mel_filters(
+        sr: int, n_fft: int, n_mels: int, fmin: float = 0.0, fmax: float = None
+    ):
+        """Create a Filterbank matrix to combine FFT bins into Mel-frequency bins.
+
+        Parameters
+        ----------
+        sr : int
+            Sample rate of audio
+        n_fft : int
+            Number of FFT bins
+        n_mels : int
+            Number of mels
+        fmin : float, optional
+            Lowest frequency, in Hz, by default 0.0
+        fmax : float, optional
+            Highest frequency, by default None
+
+        Returns
+        -------
+        np.ndarray [shape=(n_mels, 1 + n_fft/2)]
+            Mel transform matrix
+        """
+        from librosa.filters import mel as librosa_mel_fn
+
+        return librosa_mel_fn(
+            sr=sr,
+            n_fft=n_fft,
+            n_mels=n_mels,
+            fmin=fmin,
+            fmax=fmax,
+        )
+
+    def mel_spectrogram(
+        self, n_mels: int = 80, mel_fmin: float = 0.0, mel_fmax: float = None, **kwargs
+    ):
+        """Computes a Mel spectrogram.
+
+        Parameters
+        ----------
+        n_mels : int, optional
+            Number of mels, by default 80
+        mel_fmin : float, optional
+            Lowest frequency, in Hz, by default 0.0
+        mel_fmax : float, optional
+            Highest frequency, by default None
+        kwargs : dict, optional
+            Keyword arguments to self.stft().
+
+        Returns
+        -------
+        torch.Tensor [shape=(batch, channels, mels, time)]
+            Mel spectrogram.
+        """
+        stft = self.stft(**kwargs)
+        magnitude = torch.abs(stft)
+
+        nf = magnitude.shape[2]
+        mel_basis = self.get_mel_filters(
+            sr=self.sample_rate,
+            n_fft=2 * (nf - 1),
+            n_mels=n_mels,
+            fmin=mel_fmin,
+            fmax=mel_fmax,
+        )
+        mel_basis = torch.from_numpy(mel_basis).to(self.device)
+
+        mel_spectrogram = magnitude.transpose(2, -1) @ mel_basis.T
+        mel_spectrogram = mel_spectrogram.transpose(-1, 2)
+        return mel_spectrogram
+
+    @staticmethod
+    @functools.lru_cache(None)
+    def get_dct(n_mfcc: int, n_mels: int, norm: str = "ortho", device: str = None):
+        """Create a discrete cosine transform (DCT) transformation matrix with shape (``n_mels``, ``n_mfcc``),
+        it can be normalized depending on norm. For more information about dct:
+        http://en.wikipedia.org/wiki/Discrete_cosine_transform#DCT-II
+
+        Parameters
+        ----------
+        n_mfcc : int
+            Number of mfccs
+        n_mels : int
+            Number of mels
+        norm   : str
+            Use "ortho" to get a orthogonal matrix or None, by default "ortho"
+        device : str, optional
+            Device to load the transformation matrix on, by default None
+
+        Returns
+        -------
+        torch.Tensor [shape=(n_mels, n_mfcc)] T
+            The dct transformation matrix.
+        """
+        from torchaudio.functional import create_dct
+
+        return create_dct(n_mfcc, n_mels, norm).to(device)
+
+    def mfcc(
+        self, n_mfcc: int = 40, n_mels: int = 80, log_offset: float = 1e-6, **kwargs
+    ):
+        """Computes mel-frequency cepstral coefficients (MFCCs).
+
+        Parameters
+        ----------
+        n_mfcc : int, optional
+            Number of mels, by default 40
+        n_mels : int, optional
+            Number of mels, by default 80
+        log_offset: float, optional
+            Small value to prevent numerical issues when trying to compute log(0), by default 1e-6
+        kwargs : dict, optional
+            Keyword arguments to self.mel_spectrogram(), note that some of them will be used for self.stft()
+
+        Returns
+        -------
+        torch.Tensor [shape=(batch, channels, mfccs, time)]
+            MFCCs.
+        """
+
+        mel_spectrogram = self.mel_spectrogram(n_mels, **kwargs)
+        mel_spectrogram = torch.log(mel_spectrogram + log_offset)
+        dct_mat = self.get_dct(n_mfcc, n_mels, "ortho", self.device)
+
+        mfcc = mel_spectrogram.transpose(-1, -2) @ dct_mat
+        mfcc = mfcc.transpose(-1, -2)
+        return mfcc
+
+    @property
+    def magnitude(self):
+        """Computes and returns the absolute value of the STFT, which
+        is the magnitude. This value can also be set to some tensor.
+        When set, ``self.stft_data`` is manipulated so that its magnitude
+        matches what this is set to, and modulated by the phase.
+
+        Returns
+        -------
+        torch.Tensor
+            Magnitude of STFT.
+
+        Examples
+        --------
+        >>> signal = AudioSignal(torch.randn(44100), 44100)
+        >>> magnitude = signal.magnitude # Computes stft if not computed
+        >>> magnitude[magnitude < magnitude.mean()] = 0
+        >>> signal.magnitude = magnitude
+        >>> signal.istft()
+        """
+        if self.stft_data is None:
+            self.stft()
+        return torch.abs(self.stft_data)
+
+    @magnitude.setter
+    def magnitude(self, value):
+        self.stft_data = value * torch.exp(1j * self.phase)
+        return
+
+    def log_magnitude(
+        self, ref_value: float = 1.0, amin: float = 1e-5, top_db: float = 80.0
+    ):
+        """Computes the log-magnitude of the spectrogram.
+
+        Parameters
+        ----------
+        ref_value : float, optional
+            The magnitude is scaled relative to ``ref``: ``20 * log10(S / ref)``.
+            Zeros in the output correspond to positions where ``S == ref``,
+            by default 1.0
+        amin : float, optional
+            Minimum threshold for ``S`` and ``ref``, by default 1e-5
+        top_db : float, optional
+            Threshold the output at ``top_db`` below the peak:
+            ``max(10 * log10(S/ref)) - top_db``, by default -80.0
+
+        Returns
+        -------
+        torch.Tensor
+            Log-magnitude spectrogram
+        """
+        magnitude = self.magnitude
+
+        amin = amin**2
+        log_spec = 10.0 * torch.log10(magnitude.pow(2).clamp(min=amin))
+        log_spec -= 10.0 * np.log10(np.maximum(amin, ref_value))
+
+        if top_db is not None:
+            log_spec = torch.maximum(log_spec, log_spec.max() - top_db)
+        return log_spec
+
+    @property
+    def phase(self):
+        """Computes and returns the phase of the STFT.
+        This value can also be set to some tensor.
+        When set, ``self.stft_data`` is manipulated so that its phase
+        matches what this is set to, we original magnitudeith th.
+
+        Returns
+        -------
+        torch.Tensor
+            Phase of STFT.
+
+        Examples
+        --------
+        >>> signal = AudioSignal(torch.randn(44100), 44100)
+        >>> phase = signal.phase # Computes stft if not computed
+        >>> phase[phase < phase.mean()] = 0
+        >>> signal.phase = phase
+        >>> signal.istft()
+        """
+        if self.stft_data is None:
+            self.stft()
+        return torch.angle(self.stft_data)
+
+    @phase.setter
+    def phase(self, value):
+        self.stft_data = self.magnitude * torch.exp(1j * value)
+        return
+
+    # Operator overloading
+    def __add__(self, other):
+        new_signal = self.clone()
+        new_signal.audio_data += util._get_value(other)
+        return new_signal
+
+    def __iadd__(self, other):
+        self.audio_data += util._get_value(other)
+        return self
+
+    def __radd__(self, other):
+        return self + other
+
+    def __sub__(self, other):
+        new_signal = self.clone()
+        new_signal.audio_data -= util._get_value(other)
+        return new_signal
+
+    def __isub__(self, other):
+        self.audio_data -= util._get_value(other)
+        return self
+
+    def __mul__(self, other):
+        new_signal = self.clone()
+        new_signal.audio_data *= util._get_value(other)
+        return new_signal
+
+    def __imul__(self, other):
+        self.audio_data *= util._get_value(other)
+        return self
+
+    def __rmul__(self, other):
+        return self * other
+
+    # Representation
+    def _info(self):
+        dur = f"{self.signal_duration:0.3f}" if self.signal_duration else "[unknown]"
+        info = {
+            "duration": f"{dur} seconds",
+            "batch_size": self.batch_size,
+            "path": self.path_to_file if self.path_to_file else "path unknown",
+            "sample_rate": self.sample_rate,
+            "num_channels": self.num_channels if self.num_channels else "[unknown]",
+            "audio_data.shape": self.audio_data.shape,
+            "stft_params": self.stft_params,
+            "device": self.device,
+        }
+
+        return info
+
+    def markdown(self):
+        """Produces a markdown representation of AudioSignal, in a markdown table.
+
+        Returns
+        -------
+        str
+            Markdown representation of AudioSignal.
+
+        Examples
+        --------
+        >>> signal = AudioSignal(torch.randn(44100), 44100)
+        >>> print(signal.markdown())
+        | Key | Value
+        |---|---
+        | duration | 1.000 seconds |
+        | batch_size | 1 |
+        | path | path unknown |
+        | sample_rate | 44100 |
+        | num_channels | 1 |
+        | audio_data.shape | torch.Size([1, 1, 44100]) |
+        | stft_params | STFTParams(window_length=2048, hop_length=512, window_type='sqrt_hann', match_stride=False) |
+        | device | cpu |
+        """
+        info = self._info()
+
+        FORMAT = "| Key | Value \n" "|---|--- \n"
+        for k, v in info.items():
+            row = f"| {k} | {v} |\n"
+            FORMAT += row
+        return FORMAT
+
+    def __str__(self):
+        info = self._info()
+
+        desc = ""
+        for k, v in info.items():
+            desc += f"{k}: {v}\n"
+        return desc
+
+    def __rich__(self):
+        from rich.table import Table
+
+        info = self._info()
+
+        table = Table(title=f"{self.__class__.__name__}")
+        table.add_column("Key", style="green")
+        table.add_column("Value", style="cyan")
+
+        for k, v in info.items():
+            table.add_row(k, str(v))
+        return table
+
+    # Comparison
+    def __eq__(self, other):
+        for k, v in list(self.__dict__.items()):
+            if torch.is_tensor(v):
+                if not torch.allclose(v, other.__dict__[k], atol=1e-6):
+                    max_error = (v - other.__dict__[k]).abs().max()
+                    print(f"Max abs error for {k}: {max_error}")
+                    return False
+        return True
+
+    # Indexing
+    def __getitem__(self, key):
+        if torch.is_tensor(key) and key.ndim == 0 and key.item() is True:
+            assert self.batch_size == 1
+            audio_data = self.audio_data
+            _loudness = self._loudness
+            stft_data = self.stft_data
+
+        elif isinstance(key, (bool, int, list, slice, tuple)) or (
+            torch.is_tensor(key) and key.ndim <= 1
+        ):
+            # Indexing only on the batch dimension.
+            # Then let's copy over relevant stuff.
+            # Future work: make this work for time-indexing
+            # as well, using the hop length.
+            audio_data = self.audio_data[key]
+            _loudness = self._loudness[key] if self._loudness is not None else None
+            stft_data = self.stft_data[key] if self.stft_data is not None else None
+
+        sources = None
+
+        copy = type(self)(audio_data, self.sample_rate, stft_params=self.stft_params)
+        copy._loudness = _loudness
+        copy._stft_data = stft_data
+        copy.sources = sources
+
+        return copy
+
+    def __setitem__(self, key, value):
+        if not isinstance(value, type(self)):
+            self.audio_data[key] = value
+            return
+
+        if torch.is_tensor(key) and key.ndim == 0 and key.item() is True:
+            assert self.batch_size == 1
+            self.audio_data = value.audio_data
+            self._loudness = value._loudness
+            self.stft_data = value.stft_data
+            return
+
+        elif isinstance(key, (bool, int, list, slice, tuple)) or (
+            torch.is_tensor(key) and key.ndim <= 1
+        ):
+            if self.audio_data is not None and value.audio_data is not None:
+                self.audio_data[key] = value.audio_data
+            if self._loudness is not None and value._loudness is not None:
+                self._loudness[key] = value._loudness
+            if self.stft_data is not None and value.stft_data is not None:
+                self.stft_data[key] = value.stft_data
+            return
+
+    def __ne__(self, other):
+        return not self == other

dac-vae/audiotools/core/display.py

@@ -0,0 +1,194 @@
+import inspect
+import typing
+from functools import wraps
+
+from . import util
+
+
+def format_figure(func):
+    """Decorator for formatting figures produced by the code below.
+    See :py:func:`audiotools.core.util.format_figure` for more.
+
+    Parameters
+    ----------
+    func : Callable
+        Plotting function that is decorated by this function.
+
+    """
+
+    @wraps(func)
+    def wrapper(*args, **kwargs):
+        f_keys = inspect.signature(util.format_figure).parameters.keys()
+        f_kwargs = {}
+        for k, v in list(kwargs.items()):
+            if k in f_keys:
+                kwargs.pop(k)
+                f_kwargs[k] = v
+        func(*args, **kwargs)
+        util.format_figure(**f_kwargs)
+
+    return wrapper
+
+
+class DisplayMixin:
+    @format_figure
+    def specshow(
+        self,
+        preemphasis: bool = False,
+        x_axis: str = "time",
+        y_axis: str = "linear",
+        n_mels: int = 128,
+        **kwargs,
+    ):
+        """Displays a spectrogram, using ``librosa.display.specshow``.
+
+        Parameters
+        ----------
+        preemphasis : bool, optional
+            Whether or not to apply preemphasis, which makes high
+            frequency detail easier to see, by default False
+        x_axis : str, optional
+            How to label the x axis, by default "time"
+        y_axis : str, optional
+            How to label the y axis, by default "linear"
+        n_mels : int, optional
+            If displaying a mel spectrogram with ``y_axis = "mel"``,
+            this controls the number of mels, by default 128.
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.util.format_figure`.
+        """
+        import librosa
+        import librosa.display
+
+        # Always re-compute the STFT data before showing it, in case
+        # it changed.
+        signal = self.clone()
+        signal.stft_data = None
+
+        if preemphasis:
+            signal.preemphasis()
+
+        ref = signal.magnitude.max()
+        log_mag = signal.log_magnitude(ref_value=ref)
+
+        if y_axis == "mel":
+            log_mag = 20 * signal.mel_spectrogram(n_mels).clamp(1e-5).log10()
+            log_mag -= log_mag.max()
+
+        librosa.display.specshow(
+            log_mag.numpy()[0].mean(axis=0),
+            x_axis=x_axis,
+            y_axis=y_axis,
+            sr=signal.sample_rate,
+            **kwargs,
+        )
+
+    @format_figure
+    def waveplot(self, x_axis: str = "time", **kwargs):
+        """Displays a waveform plot, using ``librosa.display.waveshow``.
+
+        Parameters
+        ----------
+        x_axis : str, optional
+            How to label the x axis, by default "time"
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.util.format_figure`.
+        """
+        import librosa
+        import librosa.display
+
+        audio_data = self.audio_data[0].mean(dim=0)
+        audio_data = audio_data.cpu().numpy()
+
+        plot_fn = "waveshow" if hasattr(librosa.display, "waveshow") else "waveplot"
+        wave_plot_fn = getattr(librosa.display, plot_fn)
+        wave_plot_fn(audio_data, x_axis=x_axis, sr=self.sample_rate, **kwargs)
+
+    @format_figure
+    def wavespec(self, x_axis: str = "time", **kwargs):
+        """Displays a waveform plot, using ``librosa.display.waveshow``.
+
+        Parameters
+        ----------
+        x_axis : str, optional
+            How to label the x axis, by default "time"
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.display.DisplayMixin.specshow`.
+        """
+        import matplotlib.pyplot as plt
+        from matplotlib.gridspec import GridSpec
+
+        gs = GridSpec(6, 1)
+        plt.subplot(gs[0, :])
+        self.waveplot(x_axis=x_axis)
+        plt.subplot(gs[1:, :])
+        self.specshow(x_axis=x_axis, **kwargs)
+
+    def write_audio_to_tb(
+        self,
+        tag: str,
+        writer,
+        step: int = None,
+        plot_fn: typing.Union[typing.Callable, str] = "specshow",
+        **kwargs,
+    ):
+        """Writes a signal and its spectrogram to Tensorboard. Will show up
+        under the Audio and Images tab in Tensorboard.
+
+        Parameters
+        ----------
+        tag : str
+            Tag to write signal to (e.g. ``clean/sample_0.wav``). The image will be
+            written to the corresponding ``.png`` file (e.g. ``clean/sample_0.png``).
+        writer : SummaryWriter
+            A SummaryWriter object from PyTorch library.
+        step : int, optional
+            The step to write the signal to, by default None
+        plot_fn : typing.Union[typing.Callable, str], optional
+            How to create the image. Set to ``None`` to avoid plotting, by default "specshow"
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.display.DisplayMixin.specshow` or
+            whatever ``plot_fn`` is set to.
+        """
+        import matplotlib.pyplot as plt
+
+        audio_data = self.audio_data[0, 0].detach().cpu()
+        sample_rate = self.sample_rate
+        writer.add_audio(tag, audio_data, step, sample_rate)
+
+        if plot_fn is not None:
+            if isinstance(plot_fn, str):
+                plot_fn = getattr(self, plot_fn)
+            fig = plt.figure()
+            plt.clf()
+            plot_fn(**kwargs)
+            writer.add_figure(tag.replace("wav", "png"), fig, step)
+
+    def save_image(
+        self,
+        image_path: str,
+        plot_fn: typing.Union[typing.Callable, str] = "specshow",
+        **kwargs,
+    ):
+        """Save AudioSignal spectrogram (or whatever ``plot_fn`` is set to) to
+        a specified file.
+
+        Parameters
+        ----------
+        image_path : str
+            Where to save the file to.
+        plot_fn : typing.Union[typing.Callable, str], optional
+            How to create the image. Set to ``None`` to avoid plotting, by default "specshow"
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.display.DisplayMixin.specshow` or
+            whatever ``plot_fn`` is set to.
+        """
+        import matplotlib.pyplot as plt
+
+        if isinstance(plot_fn, str):
+            plot_fn = getattr(self, plot_fn)
+
+        plt.clf()
+        plot_fn(**kwargs)
+        plt.savefig(image_path, bbox_inches="tight", pad_inches=0)
+        plt.close()

dac-vae/audiotools/core/dsp.py

@@ -0,0 +1,390 @@
+import typing
+
+import julius
+import numpy as np
+import torch
+
+from . import util
+
+
+class DSPMixin:
+    _original_batch_size = None
+    _original_num_channels = None
+    _padded_signal_length = None
+
+    def _preprocess_signal_for_windowing(self, window_duration, hop_duration):
+        self._original_batch_size = self.batch_size
+        self._original_num_channels = self.num_channels
+
+        window_length = int(window_duration * self.sample_rate)
+        hop_length = int(hop_duration * self.sample_rate)
+
+        if window_length % hop_length != 0:
+            factor = window_length // hop_length
+            window_length = factor * hop_length
+
+        self.zero_pad(hop_length, hop_length)
+        self._padded_signal_length = self.signal_length
+
+        return window_length, hop_length
+
+    def windows(
+        self, window_duration: float, hop_duration: float, preprocess: bool = True
+    ):
+        """Generator which yields windows of specified duration from signal with a specified
+        hop length.
+
+        Parameters
+        ----------
+        window_duration : float
+            Duration of every window in seconds.
+        hop_duration : float
+            Hop between windows in seconds.
+        preprocess : bool, optional
+            Whether to preprocess the signal, so that the first sample is in
+            the middle of the first window, by default True
+
+        Yields
+        ------
+        AudioSignal
+            Each window is returned as an AudioSignal.
+        """
+        if preprocess:
+            window_length, hop_length = self._preprocess_signal_for_windowing(
+                window_duration, hop_duration
+            )
+
+        self.audio_data = self.audio_data.reshape(-1, 1, self.signal_length)
+
+        for b in range(self.batch_size):
+            i = 0
+            start_idx = i * hop_length
+            while True:
+                start_idx = i * hop_length
+                i += 1
+                end_idx = start_idx + window_length
+                if end_idx > self.signal_length:
+                    break
+                yield self[b, ..., start_idx:end_idx]
+
+    def collect_windows(
+        self, window_duration: float, hop_duration: float, preprocess: bool = True
+    ):
+        """Reshapes signal into windows of specified duration from signal with a specified
+        hop length. Window are placed along the batch dimension. Use with
+        :py:func:`audiotools.core.dsp.DSPMixin.overlap_and_add` to reconstruct the
+        original signal.
+
+        Parameters
+        ----------
+        window_duration : float
+            Duration of every window in seconds.
+        hop_duration : float
+            Hop between windows in seconds.
+        preprocess : bool, optional
+            Whether to preprocess the signal, so that the first sample is in
+            the middle of the first window, by default True
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal unfolded with shape ``(nb * nch * num_windows, 1, window_length)``
+        """
+        if preprocess:
+            window_length, hop_length = self._preprocess_signal_for_windowing(
+                window_duration, hop_duration
+            )
+
+        # self.audio_data: (nb, nch, nt).
+        unfolded = torch.nn.functional.unfold(
+            self.audio_data.reshape(-1, 1, 1, self.signal_length),
+            kernel_size=(1, window_length),
+            stride=(1, hop_length),
+        )
+        # unfolded: (nb * nch, window_length, num_windows).
+        # -> (nb * nch * num_windows, 1, window_length)
+        unfolded = unfolded.permute(0, 2, 1).reshape(-1, 1, window_length)
+        self.audio_data = unfolded
+        return self
+
+    def overlap_and_add(self, hop_duration: float):
+        """Function which takes a list of windows and overlap adds them into a
+        signal the same length as ``audio_signal``.
+
+        Parameters
+        ----------
+        hop_duration : float
+            How much to shift for each window
+            (overlap is window_duration - hop_duration) in seconds.
+
+        Returns
+        -------
+        AudioSignal
+            overlap-and-added signal.
+        """
+        hop_length = int(hop_duration * self.sample_rate)
+        window_length = self.signal_length
+
+        nb, nch = self._original_batch_size, self._original_num_channels
+
+        unfolded = self.audio_data.reshape(nb * nch, -1, window_length).permute(0, 2, 1)
+        folded = torch.nn.functional.fold(
+            unfolded,
+            output_size=(1, self._padded_signal_length),
+            kernel_size=(1, window_length),
+            stride=(1, hop_length),
+        )
+
+        norm = torch.ones_like(unfolded, device=unfolded.device)
+        norm = torch.nn.functional.fold(
+            norm,
+            output_size=(1, self._padded_signal_length),
+            kernel_size=(1, window_length),
+            stride=(1, hop_length),
+        )
+
+        folded = folded / norm
+
+        folded = folded.reshape(nb, nch, -1)
+        self.audio_data = folded
+        self.trim(hop_length, hop_length)
+        return self
+
+    def low_pass(
+        self, cutoffs: typing.Union[torch.Tensor, np.ndarray, float], zeros: int = 51
+    ):
+        """Low-passes the signal in-place. Each item in the batch
+        can have a different low-pass cutoff, if the input
+        to this signal is an array or tensor. If a float, all
+        items are given the same low-pass filter.
+
+        Parameters
+        ----------
+        cutoffs : typing.Union[torch.Tensor, np.ndarray, float]
+            Cutoff in Hz of low-pass filter.
+        zeros : int, optional
+            Number of taps to use in low-pass filter, by default 51
+
+        Returns
+        -------
+        AudioSignal
+            Low-passed AudioSignal.
+        """
+        cutoffs = util.ensure_tensor(cutoffs, 2, self.batch_size)
+        cutoffs = cutoffs / self.sample_rate
+        filtered = torch.empty_like(self.audio_data)
+
+        for i, cutoff in enumerate(cutoffs):
+            lp_filter = julius.LowPassFilter(cutoff.cpu(), zeros=zeros).to(self.device)
+            filtered[i] = lp_filter(self.audio_data[i])
+
+        self.audio_data = filtered
+        self.stft_data = None
+        return self
+
+    def high_pass(
+        self, cutoffs: typing.Union[torch.Tensor, np.ndarray, float], zeros: int = 51
+    ):
+        """High-passes the signal in-place. Each item in the batch
+        can have a different high-pass cutoff, if the input
+        to this signal is an array or tensor. If a float, all
+        items are given the same high-pass filter.
+
+        Parameters
+        ----------
+        cutoffs : typing.Union[torch.Tensor, np.ndarray, float]
+            Cutoff in Hz of high-pass filter.
+        zeros : int, optional
+            Number of taps to use in high-pass filter, by default 51
+
+        Returns
+        -------
+        AudioSignal
+            High-passed AudioSignal.
+        """
+        cutoffs = util.ensure_tensor(cutoffs, 2, self.batch_size)
+        cutoffs = cutoffs / self.sample_rate
+        filtered = torch.empty_like(self.audio_data)
+
+        for i, cutoff in enumerate(cutoffs):
+            hp_filter = julius.HighPassFilter(cutoff.cpu(), zeros=zeros).to(self.device)
+            filtered[i] = hp_filter(self.audio_data[i])
+
+        self.audio_data = filtered
+        self.stft_data = None
+        return self
+
+    def mask_frequencies(
+        self,
+        fmin_hz: typing.Union[torch.Tensor, np.ndarray, float],
+        fmax_hz: typing.Union[torch.Tensor, np.ndarray, float],
+        val: float = 0.0,
+    ):
+        """Masks frequencies between ``fmin_hz`` and ``fmax_hz``, and fills them
+        with the value specified by ``val``. Useful for implementing SpecAug.
+        The min and max can be different for every item in the batch.
+
+        Parameters
+        ----------
+        fmin_hz : typing.Union[torch.Tensor, np.ndarray, float]
+            Lower end of band to mask out.
+        fmax_hz : typing.Union[torch.Tensor, np.ndarray, float]
+            Upper end of band to mask out.
+        val : float, optional
+            Value to fill in, by default 0.0
+
+        Returns
+        -------
+        AudioSignal
+            Signal with ``stft_data`` manipulated. Apply ``.istft()`` to get the
+            masked audio data.
+        """
+        # SpecAug
+        mag, phase = self.magnitude, self.phase
+        fmin_hz = util.ensure_tensor(fmin_hz, ndim=mag.ndim)
+        fmax_hz = util.ensure_tensor(fmax_hz, ndim=mag.ndim)
+        assert torch.all(fmin_hz < fmax_hz)
+
+        # build mask
+        nbins = mag.shape[-2]
+        bins_hz = torch.linspace(0, self.sample_rate / 2, nbins, device=self.device)
+        bins_hz = bins_hz[None, None, :, None].repeat(
+            self.batch_size, 1, 1, mag.shape[-1]
+        )
+        mask = (fmin_hz <= bins_hz) & (bins_hz < fmax_hz)
+        mask = mask.to(self.device)
+
+        mag = mag.masked_fill(mask, val)
+        phase = phase.masked_fill(mask, val)
+        self.stft_data = mag * torch.exp(1j * phase)
+        return self
+
+    def mask_timesteps(
+        self,
+        tmin_s: typing.Union[torch.Tensor, np.ndarray, float],
+        tmax_s: typing.Union[torch.Tensor, np.ndarray, float],
+        val: float = 0.0,
+    ):
+        """Masks timesteps between ``tmin_s`` and ``tmax_s``, and fills them
+        with the value specified by ``val``. Useful for implementing SpecAug.
+        The min and max can be different for every item in the batch.
+
+        Parameters
+        ----------
+        tmin_s : typing.Union[torch.Tensor, np.ndarray, float]
+            Lower end of timesteps to mask out.
+        tmax_s : typing.Union[torch.Tensor, np.ndarray, float]
+            Upper end of timesteps to mask out.
+        val : float, optional
+            Value to fill in, by default 0.0
+
+        Returns
+        -------
+        AudioSignal
+            Signal with ``stft_data`` manipulated. Apply ``.istft()`` to get the
+            masked audio data.
+        """
+        # SpecAug
+        mag, phase = self.magnitude, self.phase
+        tmin_s = util.ensure_tensor(tmin_s, ndim=mag.ndim)
+        tmax_s = util.ensure_tensor(tmax_s, ndim=mag.ndim)
+
+        assert torch.all(tmin_s < tmax_s)
+
+        # build mask
+        nt = mag.shape[-1]
+        bins_t = torch.linspace(0, self.signal_duration, nt, device=self.device)
+        bins_t = bins_t[None, None, None, :].repeat(
+            self.batch_size, 1, mag.shape[-2], 1
+        )
+        mask = (tmin_s <= bins_t) & (bins_t < tmax_s)
+
+        mag = mag.masked_fill(mask, val)
+        phase = phase.masked_fill(mask, val)
+        self.stft_data = mag * torch.exp(1j * phase)
+        return self
+
+    def mask_low_magnitudes(
+        self, db_cutoff: typing.Union[torch.Tensor, np.ndarray, float], val: float = 0.0
+    ):
+        """Mask away magnitudes below a specified threshold, which
+        can be different for every item in the batch.
+
+        Parameters
+        ----------
+        db_cutoff : typing.Union[torch.Tensor, np.ndarray, float]
+            Decibel value for which things below it will be masked away.
+        val : float, optional
+            Value to fill in for masked portions, by default 0.0
+
+        Returns
+        -------
+        AudioSignal
+            Signal with ``stft_data`` manipulated. Apply ``.istft()`` to get the
+            masked audio data.
+        """
+        mag = self.magnitude
+        log_mag = self.log_magnitude()
+
+        db_cutoff = util.ensure_tensor(db_cutoff, ndim=mag.ndim)
+        mask = log_mag < db_cutoff
+        mag = mag.masked_fill(mask, val)
+
+        self.magnitude = mag
+        return self
+
+    def shift_phase(self, shift: typing.Union[torch.Tensor, np.ndarray, float]):
+        """Shifts the phase by a constant value.
+
+        Parameters
+        ----------
+        shift : typing.Union[torch.Tensor, np.ndarray, float]
+            What to shift the phase by.
+
+        Returns
+        -------
+        AudioSignal
+            Signal with ``stft_data`` manipulated. Apply ``.istft()`` to get the
+            masked audio data.
+        """
+        shift = util.ensure_tensor(shift, ndim=self.phase.ndim)
+        self.phase = self.phase + shift
+        return self
+
+    def corrupt_phase(self, scale: typing.Union[torch.Tensor, np.ndarray, float]):
+        """Corrupts the phase randomly by some scaled value.
+
+        Parameters
+        ----------
+        scale : typing.Union[torch.Tensor, np.ndarray, float]
+            Standard deviation of noise to add to the phase.
+
+        Returns
+        -------
+        AudioSignal
+            Signal with ``stft_data`` manipulated. Apply ``.istft()`` to get the
+            masked audio data.
+        """
+        scale = util.ensure_tensor(scale, ndim=self.phase.ndim)
+        self.phase = self.phase + scale * torch.randn_like(self.phase)
+        return self
+
+    def preemphasis(self, coef: float = 0.85):
+        """Applies pre-emphasis to audio signal.
+
+        Parameters
+        ----------
+        coef : float, optional
+            How much pre-emphasis to apply, lower values do less. 0 does nothing.
+            by default 0.85
+
+        Returns
+        -------
+        AudioSignal
+            Pre-emphasized signal.
+        """
+        kernel = torch.tensor([1, -coef, 0]).view(1, 1, -1).to(self.device)
+        x = self.audio_data.reshape(-1, 1, self.signal_length)
+        x = torch.nn.functional.conv1d(x, kernel, padding=1)
+        self.audio_data = x.reshape(*self.audio_data.shape)
+        return self

dac-vae/audiotools/core/effects.py

@@ -0,0 +1,647 @@
+import typing
+
+import julius
+import numpy as np
+import torch
+import torchaudio
+
+from . import util
+
+
+class EffectMixin:
+    GAIN_FACTOR = np.log(10) / 20
+    """Gain factor for converting between amplitude and decibels."""
+    CODEC_PRESETS = {
+        "8-bit": {"format": "wav", "encoding": "ULAW", "bits_per_sample": 8},
+        "GSM-FR": {"format": "gsm"},
+        "MP3": {"format": "mp3", "compression": -9},
+        "Vorbis": {"format": "vorbis", "compression": -1},
+        "Ogg": {
+            "format": "ogg",
+            "compression": -1,
+        },
+        "Amr-nb": {"format": "amr-nb"},
+    }
+    """Presets for applying codecs via torchaudio."""
+
+    def mix(
+        self,
+        other,
+        snr: typing.Union[torch.Tensor, np.ndarray, float] = 10,
+        other_eq: typing.Union[torch.Tensor, np.ndarray] = None,
+    ):
+        """Mixes noise with signal at specified
+        signal-to-noise ratio. Optionally, the
+        other signal can be equalized in-place.
+
+
+        Parameters
+        ----------
+        other : AudioSignal
+            AudioSignal object to mix with.
+        snr : typing.Union[torch.Tensor, np.ndarray, float], optional
+            Signal to noise ratio, by default 10
+        other_eq : typing.Union[torch.Tensor, np.ndarray], optional
+            EQ curve to apply to other signal, if any, by default None
+
+        Returns
+        -------
+        AudioSignal
+            In-place modification of AudioSignal.
+        """
+        snr = util.ensure_tensor(snr).to(self.device)
+
+        pad_len = max(0, self.signal_length - other.signal_length)
+        other.zero_pad(0, pad_len)
+        other.truncate_samples(self.signal_length)
+        if other_eq is not None:
+            other = other.equalizer(other_eq)
+
+        tgt_loudness = self.loudness() - snr
+        other = other.normalize(tgt_loudness)
+
+        self.audio_data = self.audio_data + other.audio_data
+        return self
+
+    def convolve(self, other, start_at_max: bool = True):
+        """Convolves self with other.
+        This function uses FFTs to do the convolution.
+
+        Parameters
+        ----------
+        other : AudioSignal
+            Signal to convolve with.
+        start_at_max : bool, optional
+            Whether to start at the max value of other signal, to
+            avoid inducing delays, by default True
+
+        Returns
+        -------
+        AudioSignal
+            Convolved signal, in-place.
+        """
+        from . import AudioSignal
+
+        pad_len = self.signal_length - other.signal_length
+
+        if pad_len > 0:
+            other.zero_pad(0, pad_len)
+        else:
+            other.truncate_samples(self.signal_length)
+
+        if start_at_max:
+            # Use roll to rotate over the max for every item
+            # so that the impulse responses don't induce any
+            # delay.
+            idx = other.audio_data.abs().argmax(axis=-1)
+            irs = torch.zeros_like(other.audio_data)
+            for i in range(other.batch_size):
+                irs[i] = torch.roll(other.audio_data[i], -idx[i].item(), -1)
+            other = AudioSignal(irs, other.sample_rate)
+
+        delta = torch.zeros_like(other.audio_data)
+        delta[..., 0] = 1
+
+        length = self.signal_length
+        delta_fft = torch.fft.rfft(delta, length)
+        other_fft = torch.fft.rfft(other.audio_data, length)
+        self_fft = torch.fft.rfft(self.audio_data, length)
+
+        convolved_fft = other_fft * self_fft
+        convolved_audio = torch.fft.irfft(convolved_fft, length)
+
+        delta_convolved_fft = other_fft * delta_fft
+        delta_audio = torch.fft.irfft(delta_convolved_fft, length)
+
+        # Use the delta to rescale the audio exactly as needed.
+        delta_max = delta_audio.abs().max(dim=-1, keepdims=True)[0]
+        scale = 1 / delta_max.clamp(1e-5)
+        convolved_audio = convolved_audio * scale
+
+        self.audio_data = convolved_audio
+
+        return self
+
+    def apply_ir(
+        self,
+        ir,
+        drr: typing.Union[torch.Tensor, np.ndarray, float] = None,
+        ir_eq: typing.Union[torch.Tensor, np.ndarray] = None,
+        use_original_phase: bool = False,
+    ):
+        """Applies an impulse response to the signal. If ` is`ir_eq``
+        is specified, the impulse response is equalized before
+        it is applied, using the given curve.
+
+        Parameters
+        ----------
+        ir : AudioSignal
+            Impulse response to convolve with.
+        drr : typing.Union[torch.Tensor, np.ndarray, float], optional
+            Direct-to-reverberant ratio that impulse response will be
+            altered to, if specified, by default None
+        ir_eq : typing.Union[torch.Tensor, np.ndarray], optional
+            Equalization that will be applied to impulse response
+            if specified, by default None
+        use_original_phase : bool, optional
+            Whether to use the original phase, instead of the convolved
+            phase, by default False
+
+        Returns
+        -------
+        AudioSignal
+            Signal with impulse response applied to it
+        """
+        if ir_eq is not None:
+            ir = ir.equalizer(ir_eq)
+        if drr is not None:
+            ir = ir.alter_drr(drr)
+
+        # Save the peak before
+        max_spk = self.audio_data.abs().max(dim=-1, keepdims=True).values
+
+        # Augment the impulse response to simulate microphone effects
+        # and with varying direct-to-reverberant ratio.
+        phase = self.phase
+        self.convolve(ir)
+
+        # Use the input phase
+        if use_original_phase:
+            self.stft()
+            self.stft_data = self.magnitude * torch.exp(1j * phase)
+            self.istft()
+
+        # Rescale to the input's amplitude
+        max_transformed = self.audio_data.abs().max(dim=-1, keepdims=True).values
+        scale_factor = max_spk.clamp(1e-8) / max_transformed.clamp(1e-8)
+        self = self * scale_factor
+
+        return self
+
+    def ensure_max_of_audio(self, max: float = 1.0):
+        """Ensures that ``abs(audio_data) <= max``.
+
+        Parameters
+        ----------
+        max : float, optional
+            Max absolute value of signal, by default 1.0
+
+        Returns
+        -------
+        AudioSignal
+            Signal with values scaled between -max and max.
+        """
+        peak = self.audio_data.abs().max(dim=-1, keepdims=True)[0]
+        peak_gain = torch.ones_like(peak)
+        peak_gain[peak > max] = max / peak[peak > max]
+        self.audio_data = self.audio_data * peak_gain
+        return self
+
+    def normalize(self, db: typing.Union[torch.Tensor, np.ndarray, float] = -24.0):
+        """Normalizes the signal's volume to the specified db, in LUFS.
+        This is GPU-compatible, making for very fast loudness normalization.
+
+        Parameters
+        ----------
+        db : typing.Union[torch.Tensor, np.ndarray, float], optional
+            Loudness to normalize to, by default -24.0
+
+        Returns
+        -------
+        AudioSignal
+            Normalized audio signal.
+        """
+        db = util.ensure_tensor(db).to(self.device)
+        ref_db = self.loudness()
+        gain = db - ref_db
+        gain = torch.exp(gain * self.GAIN_FACTOR)
+
+        self.audio_data = self.audio_data * gain[:, None, None]
+        return self
+
+    def volume_change(self, db: typing.Union[torch.Tensor, np.ndarray, float]):
+        """Change volume of signal by some amount, in dB.
+
+        Parameters
+        ----------
+        db : typing.Union[torch.Tensor, np.ndarray, float]
+            Amount to change volume by.
+
+        Returns
+        -------
+        AudioSignal
+            Signal at new volume.
+        """
+        db = util.ensure_tensor(db, ndim=1).to(self.device)
+        gain = torch.exp(db * self.GAIN_FACTOR)
+        self.audio_data = self.audio_data * gain[:, None, None]
+        return self
+
+    def _to_2d(self):
+        waveform = self.audio_data.reshape(-1, self.signal_length)
+        return waveform
+
+    def _to_3d(self, waveform):
+        return waveform.reshape(self.batch_size, self.num_channels, -1)
+
+    def pitch_shift(self, n_semitones: int, quick: bool = True):
+        """Pitch shift the signal. All items in the batch
+        get the same pitch shift.
+
+        Parameters
+        ----------
+        n_semitones : int
+            How many semitones to shift the signal by.
+        quick : bool, optional
+            Using quick pitch shifting, by default True
+
+        Returns
+        -------
+        AudioSignal
+            Pitch shifted audio signal.
+        """
+        device = self.device
+        effects = [
+            ["pitch", str(n_semitones * 100)],
+            ["rate", str(self.sample_rate)],
+        ]
+        if quick:
+            effects[0].insert(1, "-q")
+
+        waveform = self._to_2d().cpu()
+        waveform, sample_rate = torchaudio.sox_effects.apply_effects_tensor(
+            waveform, self.sample_rate, effects, channels_first=True
+        )
+        self.sample_rate = sample_rate
+        self.audio_data = self._to_3d(waveform)
+        return self.to(device)
+
+    def time_stretch(self, factor: float, quick: bool = True):
+        """Time stretch the audio signal.
+
+        Parameters
+        ----------
+        factor : float
+            Factor by which to stretch the AudioSignal. Typically
+            between 0.8 and 1.2.
+        quick : bool, optional
+            Whether to use quick time stretching, by default True
+
+        Returns
+        -------
+        AudioSignal
+            Time-stretched AudioSignal.
+        """
+        device = self.device
+        effects = [
+            ["tempo", str(factor)],
+            ["rate", str(self.sample_rate)],
+        ]
+        if quick:
+            effects[0].insert(1, "-q")
+
+        waveform = self._to_2d().cpu()
+        waveform, sample_rate = torchaudio.sox_effects.apply_effects_tensor(
+            waveform, self.sample_rate, effects, channels_first=True
+        )
+        self.sample_rate = sample_rate
+        self.audio_data = self._to_3d(waveform)
+        return self.to(device)
+
+    def apply_codec(
+        self,
+        preset: str = None,
+        format: str = "wav",
+        encoding: str = None,
+        bits_per_sample: int = None,
+        compression: int = None,
+    ):  # pragma: no cover
+        """Applies an audio codec to the signal.
+
+        Parameters
+        ----------
+        preset : str, optional
+            One of the keys in ``self.CODEC_PRESETS``, by default None
+        format : str, optional
+            Format for audio codec, by default "wav"
+        encoding : str, optional
+            Encoding to use, by default None
+        bits_per_sample : int, optional
+            How many bits per sample, by default None
+        compression : int, optional
+            Compression amount of codec, by default None
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with codec applied.
+
+        Raises
+        ------
+        ValueError
+            If preset is not in ``self.CODEC_PRESETS``, an error
+            is thrown.
+        """
+        torchaudio_version_070 = "0.7" in torchaudio.__version__
+        if torchaudio_version_070:
+            return self
+
+        kwargs = {
+            "format": format,
+            "encoding": encoding,
+            "bits_per_sample": bits_per_sample,
+            "compression": compression,
+        }
+
+        if preset is not None:
+            if preset in self.CODEC_PRESETS:
+                kwargs = self.CODEC_PRESETS[preset]
+            else:
+                raise ValueError(
+                    f"Unknown preset: {preset}. "
+                    f"Known presets: {list(self.CODEC_PRESETS.keys())}"
+                )
+
+        waveform = self._to_2d()
+        if kwargs["format"] in ["vorbis", "mp3", "ogg", "amr-nb"]:
+            # Apply it in a for loop
+            augmented = torch.cat(
+                [
+                    torchaudio.functional.apply_codec(
+                        waveform[i][None, :], self.sample_rate, **kwargs
+                    )
+                    for i in range(waveform.shape[0])
+                ],
+                dim=0,
+            )
+        else:
+            augmented = torchaudio.functional.apply_codec(
+                waveform, self.sample_rate, **kwargs
+            )
+        augmented = self._to_3d(augmented)
+
+        self.audio_data = augmented
+        return self
+
+    def mel_filterbank(self, n_bands: int):
+        """Breaks signal into mel bands.
+
+        Parameters
+        ----------
+        n_bands : int
+            Number of mel bands to use.
+
+        Returns
+        -------
+        torch.Tensor
+            Mel-filtered bands, with last axis being the band index.
+        """
+        filterbank = (
+            julius.SplitBands(self.sample_rate, n_bands).float().to(self.device)
+        )
+        filtered = filterbank(self.audio_data)
+        return filtered.permute(1, 2, 3, 0)
+
+    def equalizer(self, db: typing.Union[torch.Tensor, np.ndarray]):
+        """Applies a mel-spaced equalizer to the audio signal.
+
+        Parameters
+        ----------
+        db : typing.Union[torch.Tensor, np.ndarray]
+            EQ curve to apply.
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal with equalization applied.
+        """
+        db = util.ensure_tensor(db)
+        n_bands = db.shape[-1]
+        fbank = self.mel_filterbank(n_bands)
+
+        # If there's a batch dimension, make sure it's the same.
+        if db.ndim == 2:
+            if db.shape[0] != 1:
+                assert db.shape[0] == fbank.shape[0]
+        else:
+            db = db.unsqueeze(0)
+
+        weights = (10**db).to(self.device).float()
+        fbank = fbank * weights[:, None, None, :]
+        eq_audio_data = fbank.sum(-1)
+        self.audio_data = eq_audio_data
+        return self
+
+    def clip_distortion(
+        self, clip_percentile: typing.Union[torch.Tensor, np.ndarray, float]
+    ):
+        """Clips the signal at a given percentile. The higher it is,
+        the lower the threshold for clipping.
+
+        Parameters
+        ----------
+        clip_percentile : typing.Union[torch.Tensor, np.ndarray, float]
+            Values are between 0.0 to 1.0. Typical values are 0.1 or below.
+
+        Returns
+        -------
+        AudioSignal
+            Audio signal with clipped audio data.
+        """
+        clip_percentile = util.ensure_tensor(clip_percentile, ndim=1)
+        min_thresh = torch.quantile(self.audio_data, clip_percentile / 2, dim=-1)
+        max_thresh = torch.quantile(self.audio_data, 1 - (clip_percentile / 2), dim=-1)
+
+        nc = self.audio_data.shape[1]
+        min_thresh = min_thresh[:, :nc, :]
+        max_thresh = max_thresh[:, :nc, :]
+
+        self.audio_data = self.audio_data.clamp(min_thresh, max_thresh)
+
+        return self
+
+    def quantization(
+        self, quantization_channels: typing.Union[torch.Tensor, np.ndarray, int]
+    ):
+        """Applies quantization to the input waveform.
+
+        Parameters
+        ----------
+        quantization_channels : typing.Union[torch.Tensor, np.ndarray, int]
+            Number of evenly spaced quantization channels to quantize
+            to.
+
+        Returns
+        -------
+        AudioSignal
+            Quantized AudioSignal.
+        """
+        quantization_channels = util.ensure_tensor(quantization_channels, ndim=3)
+
+        x = self.audio_data
+        x = (x + 1) / 2
+        x = x * quantization_channels
+        x = x.floor()
+        x = x / quantization_channels
+        x = 2 * x - 1
+
+        residual = (self.audio_data - x).detach()
+        self.audio_data = self.audio_data - residual
+        return self
+
+    def mulaw_quantization(
+        self, quantization_channels: typing.Union[torch.Tensor, np.ndarray, int]
+    ):
+        """Applies mu-law quantization to the input waveform.
+
+        Parameters
+        ----------
+        quantization_channels : typing.Union[torch.Tensor, np.ndarray, int]
+            Number of mu-law spaced quantization channels to quantize
+            to.
+
+        Returns
+        -------
+        AudioSignal
+            Quantized AudioSignal.
+        """
+        mu = quantization_channels - 1.0
+        mu = util.ensure_tensor(mu, ndim=3)
+
+        x = self.audio_data
+
+        # quantize
+        x = torch.sign(x) * torch.log1p(mu * torch.abs(x)) / torch.log1p(mu)
+        x = ((x + 1) / 2 * mu + 0.5).to(torch.int64)
+
+        # unquantize
+        x = (x / mu) * 2 - 1.0
+        x = torch.sign(x) * (torch.exp(torch.abs(x) * torch.log1p(mu)) - 1.0) / mu
+
+        residual = (self.audio_data - x).detach()
+        self.audio_data = self.audio_data - residual
+        return self
+
+    def __matmul__(self, other):
+        return self.convolve(other)
+
+
+class ImpulseResponseMixin:
+    """These functions are generally only used with AudioSignals that are derived
+    from impulse responses, not other sources like music or speech. These methods
+    are used to replicate the data augmentation described in [1].
+
+    1.  Bryan, Nicholas J. "Impulse response data augmentation and deep
+        neural networks for blind room acoustic parameter estimation."
+        ICASSP 2020-2020 IEEE International Conference on Acoustics,
+        Speech and Signal Processing (ICASSP). IEEE, 2020.
+    """
+
+    def decompose_ir(self):
+        """Decomposes an impulse response into early and late
+        field responses.
+        """
+        # Equations 1 and 2
+        # -----------------
+        # Breaking up into early
+        # response + late field response.
+
+        td = torch.argmax(self.audio_data, dim=-1, keepdim=True)
+        t0 = int(self.sample_rate * 0.0025)
+
+        idx = torch.arange(self.audio_data.shape[-1], device=self.device)[None, None, :]
+        idx = idx.expand(self.batch_size, -1, -1)
+        early_idx = (idx >= td - t0) * (idx <= td + t0)
+
+        early_response = torch.zeros_like(self.audio_data, device=self.device)
+        early_response[early_idx] = self.audio_data[early_idx]
+
+        late_idx = ~early_idx
+        late_field = torch.zeros_like(self.audio_data, device=self.device)
+        late_field[late_idx] = self.audio_data[late_idx]
+
+        # Equation 4
+        # ----------
+        # Decompose early response into windowed
+        # direct path and windowed residual.
+
+        window = torch.zeros_like(self.audio_data, device=self.device)
+        for idx in range(self.batch_size):
+            window_idx = early_idx[idx, 0].nonzero()
+            window[idx, ..., window_idx] = self.get_window(
+                "hann", window_idx.shape[-1], self.device
+            )
+        return early_response, late_field, window
+
+    def measure_drr(self):
+        """Measures the direct-to-reverberant ratio of the impulse
+        response.
+
+        Returns
+        -------
+        float
+            Direct-to-reverberant ratio
+        """
+        early_response, late_field, _ = self.decompose_ir()
+        num = (early_response**2).sum(dim=-1)
+        den = (late_field**2).sum(dim=-1)
+        drr = 10 * torch.log10(num / den)
+        return drr
+
+    @staticmethod
+    def solve_alpha(early_response, late_field, wd, target_drr):
+        """Used to solve for the alpha value, which is used
+        to alter the drr.
+        """
+        # Equation 5
+        # ----------
+        # Apply the good ol' quadratic formula.
+
+        wd_sq = wd**2
+        wd_sq_1 = (1 - wd) ** 2
+        e_sq = early_response**2
+        l_sq = late_field**2
+        a = (wd_sq * e_sq).sum(dim=-1)
+        b = (2 * (1 - wd) * wd * e_sq).sum(dim=-1)
+        c = (wd_sq_1 * e_sq).sum(dim=-1) - torch.pow(10, target_drr / 10) * l_sq.sum(
+            dim=-1
+        )
+
+        expr = ((b**2) - 4 * a * c).sqrt()
+        alpha = torch.maximum(
+            (-b - expr) / (2 * a),
+            (-b + expr) / (2 * a),
+        )
+        return alpha
+
+    def alter_drr(self, drr: typing.Union[torch.Tensor, np.ndarray, float]):
+        """Alters the direct-to-reverberant ratio of the impulse response.
+
+        Parameters
+        ----------
+        drr : typing.Union[torch.Tensor, np.ndarray, float]
+            Direct-to-reverberant ratio that impulse response will be
+            altered to, if specified, by default None
+
+        Returns
+        -------
+        AudioSignal
+            Altered impulse response.
+        """
+        drr = util.ensure_tensor(drr, 2, self.batch_size).to(self.device)
+
+        early_response, late_field, window = self.decompose_ir()
+        alpha = self.solve_alpha(early_response, late_field, window, drr)
+        min_alpha = (
+            late_field.abs().max(dim=-1)[0] / early_response.abs().max(dim=-1)[0]
+        )
+        alpha = torch.maximum(alpha, min_alpha)[..., None]
+
+        aug_ir_data = (
+            alpha * window * early_response
+            + ((1 - window) * early_response)
+            + late_field
+        )
+        self.audio_data = aug_ir_data
+        self.ensure_max_of_audio()
+        return self

dac-vae/audiotools/core/ffmpeg.py

@@ -0,0 +1,211 @@
+import json
+import shlex
+import subprocess
+import tempfile
+from pathlib import Path
+from typing import Tuple
+
+import ffmpy
+import numpy as np
+import torch
+
+
+def r128stats(filepath: str, quiet: bool):
+    """Takes a path to an audio file, returns a dict with the loudness
+    stats computed by the ffmpeg ebur128 filter.
+
+    Parameters
+    ----------
+    filepath : str
+        Path to compute loudness stats on.
+    quiet : bool
+        Whether to show FFMPEG output during computation.
+
+    Returns
+    -------
+    dict
+        Dictionary containing loudness stats.
+    """
+    ffargs = [
+        "ffmpeg",
+        "-nostats",
+        "-i",
+        filepath,
+        "-filter_complex",
+        "ebur128",
+        "-f",
+        "null",
+        "-",
+    ]
+    if quiet:
+        ffargs += ["-hide_banner"]
+    proc = subprocess.Popen(ffargs, stderr=subprocess.PIPE, universal_newlines=True)
+    stats = proc.communicate()[1]
+    summary_index = stats.rfind("Summary:")
+
+    summary_list = stats[summary_index:].split()
+    i_lufs = float(summary_list[summary_list.index("I:") + 1])
+    i_thresh = float(summary_list[summary_list.index("I:") + 4])
+    lra = float(summary_list[summary_list.index("LRA:") + 1])
+    lra_thresh = float(summary_list[summary_list.index("LRA:") + 4])
+    lra_low = float(summary_list[summary_list.index("low:") + 1])
+    lra_high = float(summary_list[summary_list.index("high:") + 1])
+    stats_dict = {
+        "I": i_lufs,
+        "I Threshold": i_thresh,
+        "LRA": lra,
+        "LRA Threshold": lra_thresh,
+        "LRA Low": lra_low,
+        "LRA High": lra_high,
+    }
+
+    return stats_dict
+
+
+def ffprobe_offset_and_codec(path: str) -> Tuple[float, str]:
+    """Given a path to a file, returns the start time offset and codec of
+    the first audio stream.
+    """
+    ff = ffmpy.FFprobe(
+        inputs={path: None},
+        global_options="-show_entries format=start_time:stream=duration,start_time,codec_type,codec_name,start_pts,time_base -of json -v quiet",
+    )
+    streams = json.loads(ff.run(stdout=subprocess.PIPE)[0])["streams"]
+    seconds_offset = 0.0
+    codec = None
+
+    # Get the offset and codec of the first audio stream we find
+    # and return its start time, if it has one.
+    for stream in streams:
+        if stream["codec_type"] == "audio":
+            seconds_offset = stream.get("start_time", 0.0)
+            codec = stream.get("codec_name")
+            break
+    return float(seconds_offset), codec
+
+
+class FFMPEGMixin:
+    _loudness = None
+
+    def ffmpeg_loudness(self, quiet: bool = True):
+        """Computes loudness of audio file using FFMPEG.
+
+        Parameters
+        ----------
+        quiet : bool, optional
+            Whether to show FFMPEG output during computation,
+            by default True
+
+        Returns
+        -------
+        torch.Tensor
+            Loudness of every item in the batch, computed via
+            FFMPEG.
+        """
+        loudness = []
+
+        with tempfile.NamedTemporaryFile(suffix=".wav") as f:
+            for i in range(self.batch_size):
+                self[i].write(f.name)
+                loudness_stats = r128stats(f.name, quiet=quiet)
+                loudness.append(loudness_stats["I"])
+
+        self._loudness = torch.from_numpy(np.array(loudness)).float()
+        return self.loudness()
+
+    def ffmpeg_resample(self, sample_rate: int, quiet: bool = True):
+        """Resamples AudioSignal using FFMPEG. More memory-efficient
+        than using julius.resample for long audio files.
+
+        Parameters
+        ----------
+        sample_rate : int
+            Sample rate to resample to.
+        quiet : bool, optional
+            Whether to show FFMPEG output during computation,
+            by default True
+
+        Returns
+        -------
+        AudioSignal
+            Resampled AudioSignal.
+        """
+        from audiotools import AudioSignal
+
+        if sample_rate == self.sample_rate:
+            return self
+
+        with tempfile.NamedTemporaryFile(suffix=".wav") as f:
+            self.write(f.name)
+            f_out = f.name.replace("wav", "rs.wav")
+            command = f"ffmpeg -i {f.name} -ar {sample_rate} {f_out}"
+            if quiet:
+                command += " -hide_banner -loglevel error"
+            subprocess.check_call(shlex.split(command))
+            resampled = AudioSignal(f_out)
+            Path.unlink(Path(f_out))
+        return resampled
+
+    @classmethod
+    def load_from_file_with_ffmpeg(cls, audio_path: str, quiet: bool = True, **kwargs):
+        """Loads AudioSignal object after decoding it to a wav file using FFMPEG.
+        Useful for loading audio that isn't covered by librosa's loading mechanism. Also
+        useful for loading mp3 files, without any offset.
+
+        Parameters
+        ----------
+        audio_path : str
+            Path to load AudioSignal from.
+        quiet : bool, optional
+            Whether to show FFMPEG output during computation,
+            by default True
+
+        Returns
+        -------
+        AudioSignal
+            AudioSignal loaded from file with FFMPEG.
+        """
+        audio_path = str(audio_path)
+        with tempfile.TemporaryDirectory() as d:
+            wav_file = str(Path(d) / "extracted.wav")
+            padded_wav = str(Path(d) / "padded.wav")
+
+            global_options = "-y"
+            if quiet:
+                global_options += " -loglevel error"
+
+            ff = ffmpy.FFmpeg(
+                inputs={audio_path: None},
+                # For inputs that are m4a (and others?), the input audio can
+                # have samples that don't match the sample rate. This aresample
+                # option forces ffmpeg to read timing information in the source
+                # file instead of assuming constant sample rate.
+                #
+                # This fixes an issue where an input m4a file might be a
+                # different length than the output wav file
+                outputs={wav_file: "-af aresample=async=1000"},
+                global_options=global_options,
+            )
+            ff.run()
+
+            # We pad the file using the start time offset in case it's an audio
+            # stream starting at some offset in a video container.
+            pad, codec = ffprobe_offset_and_codec(audio_path)
+
+            # For mp3s, don't pad files with discrepancies less than 0.027s -
+            # it's likely due to codec latency. The amount of latency introduced
+            # by mp3 is 1152, which is 0.0261 44khz. So we set the threshold
+            # here slightly above that.
+            # Source: https://lame.sourceforge.io/tech-FAQ.txt.
+            if codec == "mp3" and pad < 0.027:
+                pad = 0.0
+            ff = ffmpy.FFmpeg(
+                inputs={wav_file: None},
+                outputs={padded_wav: f"-af 'adelay={pad*1000}:all=true'"},
+                global_options=global_options,
+            )
+            ff.run()
+
+            signal = cls(padded_wav, **kwargs)
+
+        return signal

dac-vae/audiotools/core/loudness.py

@@ -0,0 +1,320 @@
+import copy
+
+import julius
+import numpy as np
+import scipy
+import torch
+import torch.nn.functional as F
+import torchaudio
+
+
+class Meter(torch.nn.Module):
+    """Tensorized version of pyloudnorm.Meter. Works with batched audio tensors.
+
+    Parameters
+    ----------
+    rate : int
+        Sample rate of audio.
+    filter_class : str, optional
+        Class of weighting filter used.
+        K-weighting' (default), 'Fenton/Lee 1'
+        'Fenton/Lee 2', 'Dash et al.'
+        by default "K-weighting"
+    block_size : float, optional
+        Gating block size in seconds, by default 0.400
+    zeros : int, optional
+         Number of zeros to use in FIR approximation of
+         IIR filters, by default 512
+    use_fir : bool, optional
+        Whether to use FIR approximation or exact IIR formulation.
+        If computing on GPU, ``use_fir=True`` will be used, as its
+        much faster, by default False
+    """
+
+    def __init__(
+        self,
+        rate: int,
+        filter_class: str = "K-weighting",
+        block_size: float = 0.400,
+        zeros: int = 512,
+        use_fir: bool = False,
+    ):
+        super().__init__()
+
+        self.rate = rate
+        self.filter_class = filter_class
+        self.block_size = block_size
+        self.use_fir = use_fir
+
+        G = torch.from_numpy(np.array([1.0, 1.0, 1.0, 1.41, 1.41]))
+        self.register_buffer("G", G)
+
+        # Compute impulse responses so that filtering is fast via
+        # a convolution at runtime, on GPU, unlike lfilter.
+        impulse = np.zeros((zeros,))
+        impulse[..., 0] = 1.0
+
+        firs = np.zeros((len(self._filters), 1, zeros))
+        passband_gain = torch.zeros(len(self._filters))
+
+        for i, (_, filter_stage) in enumerate(self._filters.items()):
+            firs[i] = scipy.signal.lfilter(filter_stage.b, filter_stage.a, impulse)
+            passband_gain[i] = filter_stage.passband_gain
+
+        firs = torch.from_numpy(firs[..., ::-1].copy()).float()
+
+        self.register_buffer("firs", firs)
+        self.register_buffer("passband_gain", passband_gain)
+
+    def apply_filter_gpu(self, data: torch.Tensor):
+        """Performs FIR approximation of loudness computation.
+
+        Parameters
+        ----------
+        data : torch.Tensor
+            Audio data of shape (nb, nch, nt).
+
+        Returns
+        -------
+        torch.Tensor
+            Filtered audio data.
+        """
+        # Data is of shape (nb, nch, nt)
+        # Reshape to (nb*nch, 1, nt)
+        nb, nt, nch = data.shape
+        data = data.permute(0, 2, 1)
+        data = data.reshape(nb * nch, 1, nt)
+
+        # Apply padding
+        pad_length = self.firs.shape[-1]
+
+        # Apply filtering in sequence
+        for i in range(self.firs.shape[0]):
+            data = F.pad(data, (pad_length, pad_length))
+            data = julius.fftconv.fft_conv1d(data, self.firs[i, None, ...])
+            data = self.passband_gain[i] * data
+            data = data[..., 1 : nt + 1]
+
+        data = data.permute(0, 2, 1)
+        data = data[:, :nt, :]
+        return data
+
+    def apply_filter_cpu(self, data: torch.Tensor):
+        """Performs IIR formulation of loudness computation.
+
+        Parameters
+        ----------
+        data : torch.Tensor
+            Audio data of shape (nb, nch, nt).
+
+        Returns
+        -------
+        torch.Tensor
+            Filtered audio data.
+        """
+        for _, filter_stage in self._filters.items():
+            passband_gain = filter_stage.passband_gain
+
+            a_coeffs = torch.from_numpy(filter_stage.a).float().to(data.device)
+            b_coeffs = torch.from_numpy(filter_stage.b).float().to(data.device)
+
+            _data = data.permute(0, 2, 1)
+            filtered = torchaudio.functional.lfilter(
+                _data, a_coeffs, b_coeffs, clamp=False
+            )
+            data = passband_gain * filtered.permute(0, 2, 1)
+        return data
+
+    def apply_filter(self, data: torch.Tensor):
+        """Applies filter on either CPU or GPU, depending
+        on if the audio is on GPU or is on CPU, or if
+        ``self.use_fir`` is True.
+
+        Parameters
+        ----------
+        data : torch.Tensor
+            Audio data of shape (nb, nch, nt).
+
+        Returns
+        -------
+        torch.Tensor
+            Filtered audio data.
+        """
+        if data.is_cuda or self.use_fir:
+            data = self.apply_filter_gpu(data)
+        else:
+            data = self.apply_filter_cpu(data)
+        return data
+
+    def forward(self, data: torch.Tensor):
+        """Computes integrated loudness of data.
+
+        Parameters
+        ----------
+        data : torch.Tensor
+            Audio data of shape (nb, nch, nt).
+
+        Returns
+        -------
+        torch.Tensor
+            Filtered audio data.
+        """
+        return self.integrated_loudness(data)
+
+    def _unfold(self, input_data):
+        T_g = self.block_size
+        overlap = 0.75  # overlap of 75% of the block duration
+        step = 1.0 - overlap  # step size by percentage
+
+        kernel_size = int(T_g * self.rate)
+        stride = int(T_g * self.rate * step)
+        unfolded = julius.core.unfold(input_data.permute(0, 2, 1), kernel_size, stride)
+        unfolded = unfolded.transpose(-1, -2)
+
+        return unfolded
+
+    def integrated_loudness(self, data: torch.Tensor):
+        """Computes integrated loudness of data.
+
+        Parameters
+        ----------
+        data : torch.Tensor
+            Audio data of shape (nb, nch, nt).
+
+        Returns
+        -------
+        torch.Tensor
+            Filtered audio data.
+        """
+        if not torch.is_tensor(data):
+            data = torch.from_numpy(data).float()
+        else:
+            data = data.float()
+
+        input_data = copy.copy(data)
+        # Data always has a batch and channel dimension.
+        # Is of shape (nb, nt, nch)
+        if input_data.ndim < 2:
+            input_data = input_data.unsqueeze(-1)
+        if input_data.ndim < 3:
+            input_data = input_data.unsqueeze(0)
+
+        nb, nt, nch = input_data.shape
+
+        # Apply frequency weighting filters - account
+        # for the acoustic respose of the head and auditory system
+        input_data = self.apply_filter(input_data)
+
+        G = self.G  # channel gains
+        T_g = self.block_size  # 400 ms gating block standard
+        Gamma_a = -70.0  # -70 LKFS = absolute loudness threshold
+
+        unfolded = self._unfold(input_data)
+
+        z = (1.0 / (T_g * self.rate)) * unfolded.square().sum(2)
+        l = -0.691 + 10.0 * torch.log10((G[None, :nch, None] * z).sum(1, keepdim=True))
+        l = l.expand_as(z)
+
+        # find gating block indices above absolute threshold
+        z_avg_gated = z
+        z_avg_gated[l <= Gamma_a] = 0
+        masked = l > Gamma_a
+        z_avg_gated = z_avg_gated.sum(2) / masked.sum(2)
+
+        # calculate the relative threshold value (see eq. 6)
+        Gamma_r = (
+            -0.691 + 10.0 * torch.log10((z_avg_gated * G[None, :nch]).sum(-1)) - 10.0
+        )
+        Gamma_r = Gamma_r[:, None, None]
+        Gamma_r = Gamma_r.expand(nb, nch, l.shape[-1])
+
+        # find gating block indices above relative and absolute thresholds  (end of eq. 7)
+        z_avg_gated = z
+        z_avg_gated[l <= Gamma_a] = 0
+        z_avg_gated[l <= Gamma_r] = 0
+        masked = (l > Gamma_a) * (l > Gamma_r)
+        z_avg_gated = z_avg_gated.sum(2) / masked.sum(2)
+
+        # # Cannot use nan_to_num (pytorch 1.8 does not come with GCP-supported cuda version)
+        # z_avg_gated = torch.nan_to_num(z_avg_gated)
+        z_avg_gated = torch.where(
+            z_avg_gated.isnan(), torch.zeros_like(z_avg_gated), z_avg_gated
+        )
+        z_avg_gated[z_avg_gated == float("inf")] = float(np.finfo(np.float32).max)
+        z_avg_gated[z_avg_gated == -float("inf")] = float(np.finfo(np.float32).min)
+
+        LUFS = -0.691 + 10.0 * torch.log10((G[None, :nch] * z_avg_gated).sum(1))
+        return LUFS.float()
+
+    @property
+    def filter_class(self):
+        return self._filter_class
+
+    @filter_class.setter
+    def filter_class(self, value):
+        from pyloudnorm import Meter
+
+        meter = Meter(self.rate)
+        meter.filter_class = value
+        self._filter_class = value
+        self._filters = meter._filters
+
+
+class LoudnessMixin:
+    _loudness = None
+    MIN_LOUDNESS = -70
+    """Minimum loudness possible."""
+
+    def loudness(
+        self, filter_class: str = "K-weighting", block_size: float = 0.400, **kwargs
+    ):
+        """Calculates loudness using an implementation of ITU-R BS.1770-4.
+        Allows control over gating block size and frequency weighting filters for
+        additional control. Measure the integrated gated loudness of a signal.
+
+        API is derived from PyLoudnorm, but this implementation is ported to PyTorch
+        and is tensorized across batches. When on GPU, an FIR approximation of the IIR
+        filters is used to compute loudness for speed.
+
+        Uses the weighting filters and block size defined by the meter
+        the integrated loudness is measured based upon the gating algorithm
+        defined in the ITU-R BS.1770-4 specification.
+
+        Parameters
+        ----------
+        filter_class : str, optional
+            Class of weighting filter used.
+            K-weighting' (default), 'Fenton/Lee 1'
+            'Fenton/Lee 2', 'Dash et al.'
+            by default "K-weighting"
+        block_size : float, optional
+            Gating block size in seconds, by default 0.400
+        kwargs : dict, optional
+            Keyword arguments to :py:func:`audiotools.core.loudness.Meter`.
+
+        Returns
+        -------
+        torch.Tensor
+            Loudness of audio data.
+        """
+        if self._loudness is not None:
+            return self._loudness.to(self.device)
+        original_length = self.signal_length
+        if self.signal_duration < 0.5:
+            pad_len = int((0.5 - self.signal_duration) * self.sample_rate)
+            self.zero_pad(0, pad_len)
+
+        # create BS.1770 meter
+        meter = Meter(
+            self.sample_rate, filter_class=filter_class, block_size=block_size, **kwargs
+        )
+        meter = meter.to(self.device)
+        # measure loudness
+        loudness = meter.integrated_loudness(self.audio_data.permute(0, 2, 1))
+        self.truncate_samples(original_length)
+        min_loudness = (
+            torch.ones_like(loudness, device=loudness.device) * self.MIN_LOUDNESS
+        )
+        self._loudness = torch.maximum(loudness, min_loudness)
+
+        return self._loudness.to(self.device)

dac-vae/audiotools/core/playback.py

@@ -0,0 +1,252 @@
+"""
+These are utilities that allow one to embed an AudioSignal
+as a playable object in a Jupyter notebook, or to play audio from
+the terminal, etc.
+"""  # fmt: skip
+import base64
+import io
+import random
+import string
+import subprocess
+from tempfile import NamedTemporaryFile
+
+import importlib_resources as pkg_resources
+
+from . import templates
+from .util import _close_temp_files
+from .util import format_figure
+
+headers = pkg_resources.files(templates).joinpath("headers.html").read_text()
+widget = pkg_resources.files(templates).joinpath("widget.html").read_text()
+
+DEFAULT_EXTENSION = ".wav"
+
+
+def _check_imports():  # pragma: no cover
+    try:
+        import ffmpy
+    except:
+        ffmpy = False
+
+    try:
+        import IPython
+    except:
+        raise ImportError("IPython must be installed in order to use this function!")
+    return ffmpy, IPython
+
+
+class PlayMixin:
+    def embed(self, ext: str = None, display: bool = True, return_html: bool = False):
+        """Embeds audio as a playable audio embed in a notebook, or HTML
+        document, etc.
+
+        Parameters
+        ----------
+        ext : str, optional
+            Extension to use when saving the audio, by default ".wav"
+        display : bool, optional
+            This controls whether or not to display the audio when called. This
+            is used when the embed is the last line in a Jupyter cell, to prevent
+            the audio from being embedded twice, by default True
+        return_html : bool, optional
+            Whether to return the data wrapped in an HTML audio element, by default False
+
+        Returns
+        -------
+        str
+            Either the element for display, or the HTML string of it.
+        """
+        if ext is None:
+            ext = DEFAULT_EXTENSION
+        ext = f".{ext}" if not ext.startswith(".") else ext
+        ffmpy, IPython = _check_imports()
+        sr = self.sample_rate
+        tmpfiles = []
+
+        with _close_temp_files(tmpfiles):
+            tmp_wav = NamedTemporaryFile(mode="w+", suffix=".wav", delete=False)
+            tmpfiles.append(tmp_wav)
+            self.write(tmp_wav.name)
+            if ext != ".wav" and ffmpy:
+                tmp_converted = NamedTemporaryFile(mode="w+", suffix=ext, delete=False)
+                tmpfiles.append(tmp_wav)
+                ff = ffmpy.FFmpeg(
+                    inputs={tmp_wav.name: None},
+                    outputs={
+                        tmp_converted.name: "-write_xing 0 -codec:a libmp3lame -b:a 128k -y -hide_banner -loglevel error"
+                    },
+                )
+                ff.run()
+            else:
+                tmp_converted = tmp_wav
+
+            audio_element = IPython.display.Audio(data=tmp_converted.name, rate=sr)
+            if display:
+                IPython.display.display(audio_element)
+
+        if return_html:
+            audio_element = (
+                f"<audio "
+                f"  controls "
+                f"  src='{audio_element.src_attr()}'> "
+                f"</audio> "
+            )
+        return audio_element
+
+    def widget(
+        self,
+        title: str = None,
+        ext: str = ".wav",
+        add_headers: bool = True,
+        player_width: str = "100%",
+        margin: str = "10px",
+        plot_fn: str = "specshow",
+        return_html: bool = False,
+        **kwargs,
+    ):
+        """Creates a playable widget with spectrogram. Inspired (heavily) by
+        https://sjvasquez.github.io/blog/melnet/.
+
+        Parameters
+        ----------
+        title : str, optional
+            Title of plot, placed in upper right of top-most axis.
+        ext : str, optional
+            Extension for embedding, by default ".mp3"
+        add_headers : bool, optional
+            Whether or not to add headers (use for first embed, False for later embeds), by default True
+        player_width : str, optional
+            Width of the player, as a string in a CSS rule, by default "100%"
+        margin : str, optional
+            Margin on all sides of player, by default "10px"
+        plot_fn : function, optional
+            Plotting function to use (by default self.specshow).
+        return_html : bool, optional
+            Whether to return the data wrapped in an HTML audio element, by default False
+        kwargs : dict, optional
+            Keyword arguments to plot_fn (by default self.specshow).
+
+        Returns
+        -------
+        HTML
+            HTML object.
+        """
+        import matplotlib.pyplot as plt
+
+        def _save_fig_to_tag():
+            buffer = io.BytesIO()
+
+            plt.savefig(buffer, bbox_inches="tight", pad_inches=0)
+            plt.close()
+
+            buffer.seek(0)
+            data_uri = base64.b64encode(buffer.read()).decode("ascii")
+            tag = "data:image/png;base64,{0}".format(data_uri)
+
+            return tag
+
+        _, IPython = _check_imports()
+
+        header_html = ""
+
+        if add_headers:
+            header_html = headers.replace("PLAYER_WIDTH", str(player_width))
+            header_html = header_html.replace("MARGIN", str(margin))
+            IPython.display.display(IPython.display.HTML(header_html))
+
+        widget_html = widget
+        if isinstance(plot_fn, str):
+            plot_fn = getattr(self, plot_fn)
+            kwargs["title"] = title
+        plot_fn(**kwargs)
+
+        fig = plt.gcf()
+        pixels = fig.get_size_inches() * fig.dpi
+
+        tag = _save_fig_to_tag()
+
+        # Make the source image for the levels
+        self.specshow()
+        format_figure((12, 1.5))
+        levels_tag = _save_fig_to_tag()
+
+        player_id = "".join(random.choice(string.ascii_uppercase) for _ in range(10))
+
+        audio_elem = self.embed(ext=ext, display=False)
+        widget_html = widget_html.replace("AUDIO_SRC", audio_elem.src_attr())
+        widget_html = widget_html.replace("IMAGE_SRC", tag)
+        widget_html = widget_html.replace("LEVELS_SRC", levels_tag)
+        widget_html = widget_html.replace("PLAYER_ID", player_id)
+
+        # Calculate width/height of figure based on figure size.
+        widget_html = widget_html.replace("PADDING_AMOUNT", f"{int(pixels[1])}px")
+        widget_html = widget_html.replace("MAX_WIDTH", f"{int(pixels[0])}px")
+
+        IPython.display.display(IPython.display.HTML(widget_html))
+
+        if return_html:
+            html = header_html if add_headers else ""
+            html += widget_html
+            return html
+
+    def play(self):
+        """
+        Plays an audio signal if ffplay from the ffmpeg suite of tools is installed.
+        Otherwise, will fail. The audio signal is written to a temporary file
+        and then played with ffplay.
+        """
+        tmpfiles = []
+        with _close_temp_files(tmpfiles):
+            tmp_wav = NamedTemporaryFile(suffix=".wav", delete=False)
+            tmpfiles.append(tmp_wav)
+            self.write(tmp_wav.name)
+            print(self)
+            subprocess.call(
+                [
+                    "ffplay",
+                    "-nodisp",
+                    "-autoexit",
+                    "-hide_banner",
+                    "-loglevel",
+                    "error",
+                    tmp_wav.name,
+                ]
+            )
+        return self
+
+
+if __name__ == "__main__":  # pragma: no cover
+    from audiotools import AudioSignal
+
+    signal = AudioSignal(
+        "tests/audio/spk/f10_script4_produced.mp3", offset=5, duration=5
+    )
+
+    wave_html = signal.widget(
+        "Waveform",
+        plot_fn="waveplot",
+        return_html=True,
+    )
+
+    spec_html = signal.widget("Spectrogram", return_html=True, add_headers=False)
+
+    combined_html = signal.widget(
+        "Waveform + spectrogram",
+        plot_fn="wavespec",
+        return_html=True,
+        add_headers=False,
+    )
+
+    signal.low_pass(8000)
+    lowpass_html = signal.widget(
+        "Lowpassed audio",
+        plot_fn="wavespec",
+        return_html=True,
+        add_headers=False,
+    )
+
+    with open("/tmp/index.html", "w") as f:
+        f.write(wave_html)
+        f.write(spec_html)
+        f.write(combined_html)
+        f.write(lowpass_html)

dac-vae/audiotools/core/templates/headers.html

@@ -0,0 +1,322 @@
+<style>
+    .player {
+        width: 100%;
+        /*border: 1px solid black;*/
+        margin: 10px;
+    }
+
+    .underlay img {
+        width: 100%;
+        height: 100%;
+    }
+
+    .spectrogram {
+        height: 0;
+        width: 100%;
+        position: relative;
+    }
+
+    .audio-controls {
+        width: 100%;
+        height: 54px;
+        display: flex;
+        /*border-top: 1px solid black;*/
+        /*background-color: rgb(241, 243, 244);*/
+        background-color: rgb(248, 248, 248);
+        background-color: rgb(253, 253, 254);
+        border: 1px solid rgb(205, 208, 211);
+        margin-top: 20px;
+        /*border: 1px solid black;*/
+        border-radius: 30px;
+
+    }
+
+    .play-img {
+        margin: auto;
+        height: 45%;
+        width: 45%;
+        display: block;
+    }
+
+    .download-img {
+        margin: auto;
+        height: 100%;
+        width: 100%;
+        display: block;
+    }
+
+    .pause-img {
+        margin: auto;
+        height: 45%;
+        width: 45%;
+        display: none
+    }
+
+    .playpause {
+        margin:11px 11px 11px 11px;
+        width: 32px;
+        min-width: 32px;
+        height: 32px;
+        /*background-color: rgb(241, 243, 244);*/
+        background-color: rgba(0, 0, 0, 0.0);
+        /*border-right: 1px solid black;*/
+        /*border: 1px solid red;*/
+        border-radius: 16px;
+        color: black;
+        transition: 0.25s;
+        box-sizing: border-box !important;
+    }
+
+    .download {
+        margin:11px 11px 11px 11px;
+        width: 32px;
+        min-width: 32px;
+        height: 32px;
+        /*background-color: rgb(241, 243, 244);*/
+        background-color: rgba(0, 0, 0, 0.0);
+        /*border-right: 1px solid black;*/
+        /*border: 1px solid red;*/
+        border-radius: 16px;
+        color: black;
+        transition: 0.25s;
+        box-sizing: border-box !important;
+    }
+
+    /*.playpause:disabled {
+        background-color: red;
+    }*/
+
+    .playpause:hover {
+        background-color: rgba(10, 20, 30, 0.03);
+    }
+
+    .playpause:focus {
+        outline:none;
+    }
+
+    .response {
+        padding:0px 20px 0px 0px;
+        width: calc(100% - 132px);
+        height: 100%;
+
+        /*border: 1px solid red;*/
+        /*border-bottom: 1px solid rgb(89, 89, 89);*/
+    }
+
+    .response-canvas {
+        height: 100%;
+        width: 100%;
+    }
+
+
+    .underlay {
+        height: 100%;
+        width: 100%;
+        position: absolute;
+        top: 0;
+        left: 0;
+    }
+
+    .overlay{
+        width: 0%;
+        height:100%;
+        top: 0;
+        right: 0px;
+
+        background:rgba(255, 255, 255, 0.15);
+        overflow:hidden;
+        position: absolute;
+        z-index: 10;
+        border-left: solid 1px rgba(0, 0, 0, 0.664);
+
+        position: absolute;
+        pointer-events: none;
+    }
+</style>
+
+<script>
+    !function(t){if("object"==typeof exports&&"undefined"!=typeof module)module.exports=t();else if("function"==typeof define&&define.amd)define([],t);else{("undefined"!=typeof window?window:"undefined"!=typeof global?global:"undefined"!=typeof self?self:this).pako=t()}}(function(){return function(){return function t(e,a,i){function n(s,o){if(!a[s]){if(!e[s]){var l="function"==typeof require&&require;if(!o&&l)return l(s,!0);if(r)return r(s,!0);var h=new Error("Cannot find module '"+s+"'");throw h.code="MODULE_NOT_FOUND",h}var d=a[s]={exports:{}};e[s][0].call(d.exports,function(t){return n(e[s][1][t]||t)},d,d.exports,t,e,a,i)}return a[s].exports}for(var r="function"==typeof require&&require,s=0;s<i.length;s++)n(i[s]);return n}}()({1:[function(t,e,a){"use strict";var i=t("./zlib/deflate"),n=t("./utils/common"),r=t("./utils/strings"),s=t("./zlib/messages"),o=t("./zlib/zstream"),l=Object.prototype.toString,h=0,d=-1,f=0,_=8;function u(t){if(!(this instanceof u))return new 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e,a,i,r,l,h,d,f,_,u,c=t.w_size;do{if(r=t.window_size-t.lookahead-t.strstart,t.strstart>=c+(c-T)){n.arraySet(t.window,t.window,c,c,0),t.match_start-=c,t.strstart-=c,t.block_start-=c,e=a=t.hash_size;do{i=t.head[--e],t.head[e]=i>=c?i-c:0}while(--a);e=a=c;do{i=t.prev[--e],t.prev[e]=i>=c?i-c:0}while(--a);r+=c}if(0===t.strm.avail_in)break;if(h=t.strm,d=t.window,f=t.strstart+t.lookahead,_=r,u=void 0,(u=h.avail_in)>_&&(u=_),a=0===u?0:(h.avail_in-=u,n.arraySet(d,h.input,h.next_in,u,f),1===h.state.wrap?h.adler=s(h.adler,d,u,f):2===h.state.wrap&&(h.adler=o(h.adler,d,u,f)),h.next_in+=u,h.total_in+=u,u),t.lookahead+=a,t.lookahead+t.insert>=I)for(l=t.strstart-t.insert,t.ins_h=t.window[l],t.ins_h=(t.ins_h<<t.hash_shift^t.window[l+1])&t.hash_mask;t.insert&&(t.ins_h=(t.ins_h<<t.hash_shift^t.window[l+I-1])&t.hash_mask,t.prev[l&t.w_mask]=t.head[t.ins_h],t.head[t.ins_h]=l,l++,t.insert--,!(t.lookahead+t.insert<I)););}while(t.lookahead<T&&0!==t.strm.avail_in)}function st(t,e){for(var a,i;;){if(t.lookahead<T){if(rt(t),t.lookahead<T&&e===h)return q;if(0===t.lookahead)break}if(a=0,t.lookahead>=I&&(t.ins_h=(t.ins_h<<t.hash_shift^t.window[t.strstart+I-1])&t.hash_mask,a=t.prev[t.strstart&t.w_mask]=t.head[t.ins_h],t.head[t.ins_h]=t.strstart),0!==a&&t.strstart-a<=t.w_size-T&&(t.match_length=nt(t,a)),t.match_length>=I)if(i=r._tr_tally(t,t.strstart-t.match_start,t.match_length-I),t.lookahead-=t.match_length,t.match_length<=t.max_lazy_match&&t.lookahead>=I){t.match_length--;do{t.strstart++,t.ins_h=(t.ins_h<<t.hash_shift^t.window[t.strstart+I-1])&t.hash_mask,a=t.prev[t.strstart&t.w_mask]=t.head[t.ins_h],t.head[t.ins_h]=t.strstart}while(0!=--t.match_length);t.strstart++}else t.strstart+=t.match_length,t.match_length=0,t.ins_h=t.window[t.strstart],t.ins_h=(t.ins_h<<t.hash_shift^t.window[t.strstart+1])&t.hash_mask;else i=r._tr_tally(t,0,t.window[t.strstart]),t.lookahead--,t.strstart++;if(i&&(et(t,!1),0===t.strm.avail_out))return q}return t.insert=t.strstart<I-1?t.strstart:I-1,e===_?(et(t,!0),0===t.strm.avail_out?X:W):t.last_lit&&(et(t,!1),0===t.strm.avail_out)?q:G}function ot(t,e){for(var a,i,n;;){if(t.lookahead<T){if(rt(t),t.lookahead<T&&e===h)return q;if(0===t.lookahead)break}if(a=0,t.lookahead>=I&&(t.ins_h=(t.ins_h<<t.hash_shift^t.window[t.strstart+I-1])&t.hash_mask,a=t.prev[t.strstart&t.w_mask]=t.head[t.ins_h],t.head[t.ins_h]=t.strstart),t.prev_length=t.match_length,t.prev_match=t.match_start,t.match_length=I-1,0!==a&&t.prev_length<t.max_lazy_match&&t.strstart-a<=t.w_size-T&&(t.match_length=nt(t,a),t.match_length<=5&&(t.strategy===v||t.match_length===I&&t.strstart-t.match_start>4096)&&(t.match_length=I-1)),t.prev_length>=I&&t.match_length<=t.prev_length){n=t.strstart+t.lookahead-I,i=r._tr_tally(t,t.strstart-1-t.prev_match,t.prev_length-I),t.lookahead-=t.prev_length-1,t.prev_length-=2;do{++t.strstart<=n&&(t.ins_h=(t.ins_h<<t.hash_shift^t.window[t.strstart+I-1])&t.hash_mask,a=t.prev[t.strstart&t.w_mask]=t.head[t.ins_h],t.head[t.ins_h]=t.strstart)}while(0!=--t.prev_length);if(t.match_available=0,t.match_length=I-1,t.strstart++,i&&(et(t,!1),0===t.strm.avail_out))return q}else if(t.match_available){if((i=r._tr_tally(t,0,t.window[t.strstart-1]))&&et(t,!1),t.strstart++,t.lookahead--,0===t.strm.avail_out)return q}else t.match_available=1,t.strstart++,t.lookahead--}return t.match_available&&(i=r._tr_tally(t,0,t.window[t.strstart-1]),t.match_available=0),t.insert=t.strstart<I-1?t.strstart:I-1,e===_?(et(t,!0),0===t.strm.avail_out?X:W):t.last_lit&&(et(t,!1),0===t.strm.avail_out)?q:G}function lt(t,e,a,i,n){this.good_length=t,this.max_lazy=e,this.nice_length=a,this.max_chain=i,this.func=n}function ht(){this.strm=null,this.status=0,this.pending_buf=null,this.pending_buf_size=0,this.pending_out=0,this.pending=0,this.wrap=0,this.gzhead=null,this.gzindex=0,this.method=S,this.last_flush=-1,this.w_size=0,this.w_bits=0,this.w_mask=0,this.window=null,this.window_size=0,this.prev=null,this.head=null,this.ins_h=0,this.hash_size=0,this.hash_bits=0,this.hash_mask=0,this.hash_shift=0,this.block_start=0,this.match_length=0,this.prev_match=0,this.match_available=0,this.strstart=0,this.match_start=0,this.lookahead=0,this.prev_length=0,this.max_chain_length=0,this.max_lazy_match=0,this.level=0,this.strategy=0,this.good_match=0,this.nice_match=0,this.dyn_ltree=new n.Buf16(2*O),this.dyn_dtree=new n.Buf16(2*(2*C+1)),this.bl_tree=new n.Buf16(2*(2*N+1)),$(this.dyn_ltree),$(this.dyn_dtree),$(this.bl_tree),this.l_desc=null,this.d_desc=null,this.bl_desc=null,this.bl_count=new n.Buf16(D+1),this.heap=new n.Buf16(2*R+1),$(this.heap),this.heap_len=0,this.heap_max=0,this.depth=new n.Buf16(2*R+1),$(this.depth),this.l_buf=0,this.lit_bufsize=0,this.last_lit=0,this.d_buf=0,this.opt_len=0,this.static_len=0,this.matches=0,this.insert=0,this.bi_buf=0,this.bi_valid=0}function dt(t){var e;return t&&t.state?(t.total_in=t.total_out=0,t.data_type=B,(e=t.state).pending=0,e.pending_out=0,e.wrap<0&&(e.wrap=-e.wrap),e.status=e.wrap?L:P,t.adler=2===e.wrap?0:1,e.last_flush=h,r._tr_init(e),c):Q(t,g)}function ft(t){var e,a=dt(t);return a===c&&((e=t.state).window_size=2*e.w_size,$(e.head),e.max_lazy_match=i[e.level].max_lazy,e.good_match=i[e.level].good_length,e.nice_match=i[e.level].nice_length,e.max_chain_length=i[e.level].max_chain,e.strstart=0,e.block_start=0,e.lookahead=0,e.insert=0,e.match_length=e.prev_length=I-1,e.match_available=0,e.ins_h=0),a}function _t(t,e,a,i,r,s){if(!t)return g;var o=1;if(e===p&&(e=6),i<0?(o=0,i=-i):i>15&&(o=2,i-=16),r<1||r>E||a!==S||i<8||i>15||e<0||e>9||s<0||s>x)return Q(t,g);8===i&&(i=9);var l=new ht;return t.state=l,l.strm=t,l.wrap=o,l.gzhead=null,l.w_bits=i,l.w_size=1<<l.w_bits,l.w_mask=l.w_size-1,l.hash_bits=r+7,l.hash_size=1<<l.hash_bits,l.hash_mask=l.hash_size-1,l.hash_shift=~~((l.hash_bits+I-1)/I),l.window=new n.Buf8(2*l.w_size),l.head=new n.Buf16(l.hash_size),l.prev=new n.Buf16(l.w_size),l.lit_bufsize=1<<r+6,l.pending_buf_size=4*l.lit_bufsize,l.pending_buf=new n.Buf8(l.pending_buf_size),l.d_buf=1*l.lit_bufsize,l.l_buf=3*l.lit_bufsize,l.level=e,l.strategy=s,l.method=a,ft(t)}i=[new lt(0,0,0,0,function(t,e){var a=65535;for(a>t.pending_buf_size-5&&(a=t.pending_buf_size-5);;){if(t.lookahead<=1){if(rt(t),0===t.lookahead&&e===h)return q;if(0===t.lookahead)break}t.strstart+=t.lookahead,t.lookahead=0;var i=t.block_start+a;if((0===t.strstart||t.strstart>=i)&&(t.lookahead=t.strstart-i,t.strstart=i,et(t,!1),0===t.strm.avail_out))return q;if(t.strstart-t.block_start>=t.w_size-T&&(et(t,!1),0===t.strm.avail_out))return q}return t.insert=0,e===_?(et(t,!0),0===t.strm.avail_out?X:W):(t.strstart>t.block_start&&(et(t,!1),t.strm.avail_out),q)}),new lt(4,4,8,4,st),new lt(4,5,16,8,st),new lt(4,6,32,32,st),new lt(4,4,16,16,ot),new lt(8,16,32,32,ot),new lt(8,16,128,128,ot),new lt(8,32,128,256,ot),new lt(32,128,258,1024,ot),new lt(32,258,258,4096,ot)],a.deflateInit=function(t,e){return _t(t,e,S,A,Z,z)},a.deflateInit2=_t,a.deflateReset=ft,a.deflateResetKeep=dt,a.deflateSetHeader=function(t,e){return t&&t.state?2!==t.state.wrap?g:(t.state.gzhead=e,c):g},a.deflate=function(t,e){var a,n,s,l;if(!t||!t.state||e>u||e<0)return t?Q(t,g):g;if(n=t.state,!t.output||!t.input&&0!==t.avail_in||n.status===Y&&e!==_)return Q(t,0===t.avail_out?w:g);if(n.strm=t,a=n.last_flush,n.last_flush=e,n.status===L)if(2===n.wrap)t.adler=0,at(n,31),at(n,139),at(n,8),n.gzhead?(at(n,(n.gzhead.text?1:0)+(n.gzhead.hcrc?2:0)+(n.gzhead.extra?4:0)+(n.gzhead.name?8:0)+(n.gzhead.comment?16:0)),at(n,255&n.gzhead.time),at(n,n.gzhead.time>>8&255),at(n,n.gzhead.time>>16&255),at(n,n.gzhead.time>>24&255),at(n,9===n.level?2:n.strategy>=k||n.level<2?4:0),at(n,255&n.gzhead.os),n.gzhead.extra&&n.gzhead.extra.length&&(at(n,255&n.gzhead.extra.length),at(n,n.gzhead.extra.length>>8&255)),n.gzhead.hcrc&&(t.adler=o(t.adler,n.pending_buf,n.pending,0)),n.gzindex=0,n.status=H):(at(n,0),at(n,0),at(n,0),at(n,0),at(n,0),at(n,9===n.level?2:n.strategy>=k||n.level<2?4:0),at(n,J),n.status=P);else{var m=S+(n.w_bits-8<<4)<<8;m|=(n.strategy>=k||n.level<2?0:n.level<6?1:6===n.level?2:3)<<6,0!==n.strstart&&(m|=F),m+=31-m%31,n.status=P,it(n,m),0!==n.strstart&&(it(n,t.adler>>>16),it(n,65535&t.adler)),t.adler=1}if(n.status===H)if(n.gzhead.extra){for(s=n.pending;n.gzindex<(65535&n.gzhead.extra.length)&&(n.pending!==n.pending_buf_size||(n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),tt(t),s=n.pending,n.pending!==n.pending_buf_size));)at(n,255&n.gzhead.extra[n.gzindex]),n.gzindex++;n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),n.gzindex===n.gzhead.extra.length&&(n.gzindex=0,n.status=j)}else n.status=j;if(n.status===j)if(n.gzhead.name){s=n.pending;do{if(n.pending===n.pending_buf_size&&(n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),tt(t),s=n.pending,n.pending===n.pending_buf_size)){l=1;break}l=n.gzindex<n.gzhead.name.length?255&n.gzhead.name.charCodeAt(n.gzindex++):0,at(n,l)}while(0!==l);n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),0===l&&(n.gzindex=0,n.status=K)}else n.status=K;if(n.status===K)if(n.gzhead.comment){s=n.pending;do{if(n.pending===n.pending_buf_size&&(n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),tt(t),s=n.pending,n.pending===n.pending_buf_size)){l=1;break}l=n.gzindex<n.gzhead.comment.length?255&n.gzhead.comment.charCodeAt(n.gzindex++):0,at(n,l)}while(0!==l);n.gzhead.hcrc&&n.pending>s&&(t.adler=o(t.adler,n.pending_buf,n.pending-s,s)),0===l&&(n.status=M)}else n.status=M;if(n.status===M&&(n.gzhead.hcrc?(n.pending+2>n.pending_buf_size&&tt(t),n.pending+2<=n.pending_buf_size&&(at(n,255&t.adler),at(n,t.adler>>8&255),t.adler=0,n.status=P)):n.status=P),0!==n.pending){if(tt(t),0===t.avail_out)return n.last_flush=-1,c}else if(0===t.avail_in&&V(e)<=V(a)&&e!==_)return Q(t,w);if(n.status===Y&&0!==t.avail_in)return Q(t,w);if(0!==t.avail_in||0!==n.lookahead||e!==h&&n.status!==Y){var p=n.strategy===k?function(t,e){for(var a;;){if(0===t.lookahead&&(rt(t),0===t.lookahead)){if(e===h)return q;break}if(t.match_length=0,a=r._tr_tally(t,0,t.window[t.strstart]),t.lookahead--,t.strstart++,a&&(et(t,!1),0===t.strm.avail_out))return q}return t.insert=0,e===_?(et(t,!0),0===t.strm.avail_out?X:W):t.last_lit&&(et(t,!1),0===t.strm.avail_out)?q:G}(n,e):n.strategy===y?function(t,e){for(var a,i,n,s,o=t.window;;){if(t.lookahead<=U){if(rt(t),t.lookahead<=U&&e===h)return q;if(0===t.lookahead)break}if(t.match_length=0,t.lookahead>=I&&t.strstart>0&&(i=o[n=t.strstart-1])===o[++n]&&i===o[++n]&&i===o[++n]){s=t.strstart+U;do{}while(i===o[++n]&&i===o[++n]&&i===o[++n]&&i===o[++n]&&i===o[++n]&&i===o[++n]&&i===o[++n]&&i===o[++n]&&n<s);t.match_length=U-(s-n),t.match_length>t.lookahead&&(t.match_length=t.lookahead)}if(t.match_length>=I?(a=r._tr_tally(t,1,t.match_length-I),t.lookahead-=t.match_length,t.strstart+=t.match_length,t.match_length=0):(a=r._tr_tally(t,0,t.window[t.strstart]),t.lookahead--,t.strstart++),a&&(et(t,!1),0===t.strm.avail_out))return q}return t.insert=0,e===_?(et(t,!0),0===t.strm.avail_out?X:W):t.last_lit&&(et(t,!1),0===t.strm.avail_out)?q:G}(n,e):i[n.level].func(n,e);if(p!==X&&p!==W||(n.status=Y),p===q||p===X)return 0===t.avail_out&&(n.last_flush=-1),c;if(p===G&&(e===d?r._tr_align(n):e!==u&&(r._tr_stored_block(n,0,0,!1),e===f&&($(n.head),0===n.lookahead&&(n.strstart=0,n.block_start=0,n.insert=0))),tt(t),0===t.avail_out))return n.last_flush=-1,c}return e!==_?c:n.wrap<=0?b:(2===n.wrap?(at(n,255&t.adler),at(n,t.adler>>8&255),at(n,t.adler>>16&255),at(n,t.adler>>24&255),at(n,255&t.total_in),at(n,t.total_in>>8&255),at(n,t.total_in>>16&255),at(n,t.total_in>>24&255)):(it(n,t.adler>>>16),it(n,65535&t.adler)),tt(t),n.wrap>0&&(n.wrap=-n.wrap),0!==n.pending?c:b)},a.deflateEnd=function(t){var e;return t&&t.state?(e=t.state.status)!==L&&e!==H&&e!==j&&e!==K&&e!==M&&e!==P&&e!==Y?Q(t,g):(t.state=null,e===P?Q(t,m):c):g},a.deflateSetDictionary=function(t,e){var a,i,r,o,l,h,d,f,_=e.length;if(!t||!t.state)return g;if(2===(o=(a=t.state).wrap)||1===o&&a.status!==L||a.lookahead)return g;for(1===o&&(t.adler=s(t.adler,e,_,0)),a.wrap=0,_>=a.w_size&&(0===o&&($(a.head),a.strstart=0,a.block_start=0,a.insert=0),f=new n.Buf8(a.w_size),n.arraySet(f,e,_-a.w_size,a.w_size,0),e=f,_=a.w_size),l=t.avail_in,h=t.next_in,d=t.input,t.avail_in=_,t.next_in=0,t.input=e,rt(a);a.lookahead>=I;){i=a.strstart,r=a.lookahead-(I-1);do{a.ins_h=(a.ins_h<<a.hash_shift^a.window[i+I-1])&a.hash_mask,a.prev[i&a.w_mask]=a.head[a.ins_h],a.head[a.ins_h]=i,i++}while(--r);a.strstart=i,a.lookahead=I-1,rt(a)}return a.strstart+=a.lookahead,a.block_start=a.strstart,a.insert=a.lookahead,a.lookahead=0,a.match_length=a.prev_length=I-1,a.match_available=0,t.next_in=h,t.input=d,t.avail_in=l,a.wrap=o,c},a.deflateInfo="pako deflate (from Nodeca project)"},{"../utils/common":3,"./adler32":5,"./crc32":7,"./messages":13,"./trees":14}],9:[function(t,e,a){"use strict";e.exports=function(){this.text=0,this.time=0,this.xflags=0,this.os=0,this.extra=null,this.extra_len=0,this.name="",this.comment="",this.hcrc=0,this.done=!1}},{}],10:[function(t,e,a){"use strict";e.exports=function(t,e){var a,i,n,r,s,o,l,h,d,f,_,u,c,b,g,m,w,p,v,k,y,x,z,B,S;a=t.state,i=t.next_in,B=t.input,n=i+(t.avail_in-5),r=t.next_out,S=t.output,s=r-(e-t.avail_out),o=r+(t.avail_out-257),l=a.dmax,h=a.wsize,d=a.whave,f=a.wnext,_=a.window,u=a.hold,c=a.bits,b=a.lencode,g=a.distcode,m=(1<<a.lenbits)-1,w=(1<<a.distbits)-1;t:do{c<15&&(u+=B[i++]<<c,c+=8,u+=B[i++]<<c,c+=8),p=b[u&m];e:for(;;){if(u>>>=v=p>>>24,c-=v,0===(v=p>>>16&255))S[r++]=65535&p;else{if(!(16&v)){if(0==(64&v)){p=b[(65535&p)+(u&(1<<v)-1)];continue e}if(32&v){a.mode=12;break t}t.msg="invalid literal/length code",a.mode=30;break t}k=65535&p,(v&=15)&&(c<v&&(u+=B[i++]<<c,c+=8),k+=u&(1<<v)-1,u>>>=v,c-=v),c<15&&(u+=B[i++]<<c,c+=8,u+=B[i++]<<c,c+=8),p=g[u&w];a:for(;;){if(u>>>=v=p>>>24,c-=v,!(16&(v=p>>>16&255))){if(0==(64&v)){p=g[(65535&p)+(u&(1<<v)-1)];continue a}t.msg="invalid distance code",a.mode=30;break t}if(y=65535&p,c<(v&=15)&&(u+=B[i++]<<c,(c+=8)<v&&(u+=B[i++]<<c,c+=8)),(y+=u&(1<<v)-1)>l){t.msg="invalid distance too far back",a.mode=30;break 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E:if(1024&a.flags&&((ft=a.length)>rt&&(ft=rt),ft&&(a.head&&(yt=a.head.extra_len-a.length,a.head.extra||(a.head.extra=new Array(a.head.extra_len)),i.arraySet(a.head.extra,tt,at,ft,yt)),512&a.flags&&(a.check=r(a.check,tt,ft,at)),rt-=ft,at+=ft,a.length-=ft),a.length))break t;a.length=0,a.mode=A;case A:if(2048&a.flags){if(0===rt)break t;ft=0;do{yt=tt[at+ft++],a.head&&yt&&a.length<65536&&(a.head.name+=String.fromCharCode(yt))}while(yt&&ft<rt);if(512&a.flags&&(a.check=r(a.check,tt,ft,at)),rt-=ft,at+=ft,yt)break t}else a.head&&(a.head.name=null);a.length=0,a.mode=Z;case Z:if(4096&a.flags){if(0===rt)break t;ft=0;do{yt=tt[at+ft++],a.head&&yt&&a.length<65536&&(a.head.comment+=String.fromCharCode(yt))}while(yt&&ft<rt);if(512&a.flags&&(a.check=r(a.check,tt,ft,at)),rt-=ft,at+=ft,yt)break t}else a.head&&(a.head.comment=null);a.mode=R;case R:if(512&a.flags){for(;lt<16;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}if(ot!==(65535&a.check)){t.msg="header crc 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T:if(ft=a.length){if(ft>rt&&(ft=rt),ft>st&&(ft=st),0===ft)break t;i.arraySet(et,tt,at,ft,nt),rt-=ft,at+=ft,st-=ft,nt+=ft,a.length-=ft;break}a.mode=O;break;case F:for(;lt<14;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}if(a.nlen=257+(31&ot),ot>>>=5,lt-=5,a.ndist=1+(31&ot),ot>>>=5,lt-=5,a.ncode=4+(15&ot),ot>>>=4,lt-=4,a.nlen>286||a.ndist>30){t.msg="too many length or distance symbols",a.mode=Q;break}a.have=0,a.mode=L;case L:for(;a.have<a.ncode;){for(;lt<3;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}a.lens[At[a.have++]]=7&ot,ot>>>=3,lt-=3}for(;a.have<19;)a.lens[At[a.have++]]=0;if(a.lencode=a.lendyn,a.lenbits=7,zt={bits:a.lenbits},xt=o(l,a.lens,0,19,a.lencode,0,a.work,zt),a.lenbits=zt.bits,xt){t.msg="invalid code lengths set",a.mode=Q;break}a.have=0,a.mode=H;case H:for(;a.have<a.nlen+a.ndist;){for(;mt=(St=a.lencode[ot&(1<<a.lenbits)-1])>>>16&255,wt=65535&St,!((gt=St>>>24)<=lt);){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}if(wt<16)ot>>>=gt,lt-=gt,a.lens[a.have++]=wt;else{if(16===wt){for(Bt=gt+2;lt<Bt;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}if(ot>>>=gt,lt-=gt,0===a.have){t.msg="invalid bit length repeat",a.mode=Q;break}yt=a.lens[a.have-1],ft=3+(3&ot),ot>>>=2,lt-=2}else if(17===wt){for(Bt=gt+3;lt<Bt;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}lt-=gt,yt=0,ft=3+(7&(ot>>>=gt)),ot>>>=3,lt-=3}else{for(Bt=gt+7;lt<Bt;){if(0===rt)break t;rt--,ot+=tt[at++]<<lt,lt+=8}lt-=gt,yt=0,ft=11+(127&(ot>>>=gt)),ot>>>=7,lt-=7}if(a.have+ft>a.nlen+a.ndist){t.msg="invalid bit length repeat",a.mode=Q;break}for(;ft--;)a.lens[a.have++]=yt}}if(a.mode===Q)break;if(0===a.lens[256]){t.msg="invalid code -- missing end-of-block",a.mode=Q;break}if(a.lenbits=9,zt={bits:a.lenbits},xt=o(h,a.lens,0,a.nlen,a.lencode,0,a.work,zt),a.lenbits=zt.bits,xt){t.msg="invalid literal/lengths 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v=0;v<h;v++)s[v]=r[v];if(16==u)for(v=0;v<h;v++)s[v]=r[v<<1]}else if(2==c){var p=i.tabs.tRNS,b=-1,g=-1,m=-1;if(p&&(b=p[0],g=p[1],m=p[2]),8==u)for(v=0;v<a;v++){var y=3*v;s[M=v<<2]=r[y],s[M+1]=r[y+1],s[M+2]=r[y+2],s[M+3]=255,-1!=b&&r[y]==b&&r[y+1]==g&&r[y+2]==m&&(s[M+3]=0)}if(16==u)for(v=0;v<a;v++){y=6*v;s[M=v<<2]=r[y],s[M+1]=r[y+2],s[M+2]=r[y+4],s[M+3]=255,-1!=b&&d(r,y)==b&&d(r,y+2)==g&&d(r,y+4)==m&&(s[M+3]=0)}}else if(3==c){var w=i.tabs.PLTE,A=i.tabs.tRNS,U=A?A.length:0;if(1==u)for(var _=0;_<n;_++){var q=_*o,I=_*t;for(v=0;v<t;v++){var M=I+v<<2,T=3*(z=r[q+(v>>3)]>>7-((7&v)<<0)&1);s[M]=w[T],s[M+1]=w[T+1],s[M+2]=w[T+2],s[M+3]=z<U?A[z]:255}}if(2==u)for(_=0;_<n;_++)for(q=_*o,I=_*t,v=0;v<t;v++){M=I+v<<2,T=3*(z=r[q+(v>>2)]>>6-((3&v)<<1)&3);s[M]=w[T],s[M+1]=w[T+1],s[M+2]=w[T+2],s[M+3]=z<U?A[z]:255}if(4==u)for(_=0;_<n;_++)for(q=_*o,I=_*t,v=0;v<t;v++){M=I+v<<2,T=3*(z=r[q+(v>>1)]>>4-((1&v)<<2)&15);s[M]=w[T],s[M+1]=w[T+1],s[M+2]=w[T+2],s[M+3]=z<U?A[z]:255}if(8==u)for(v=0;v<a;v++){var z;M=v<<2,T=3*(z=r[v]);s[M]=w[T],s[M+1]=w[T+1],s[M+2]=w[T+2],s[M+3]=z<U?A[z]:255}}else if(4==c){if(8==u)for(v=0;v<a;v++){M=v<<2;var R=r[N=v<<1];s[M]=R,s[M+1]=R,s[M+2]=R,s[M+3]=r[N+1]}if(16==u)for(v=0;v<a;v++){var N;M=v<<2,R=r[N=v<<2];s[M]=R,s[M+1]=R,s[M+2]=R,s[M+3]=r[N+2]}}else if(0==c){b=i.tabs.tRNS?i.tabs.tRNS:-1;if(1==u)for(v=0;v<a;v++){var L=(R=255*(r[v>>3]>>7-(7&v)&1))==255*b?0:255;l[v]=L<<24|R<<16|R<<8|R}if(2==u)for(v=0;v<a;v++){L=(R=85*(r[v>>2]>>6-((3&v)<<1)&3))==85*b?0:255;l[v]=L<<24|R<<16|R<<8|R}if(4==u)for(v=0;v<a;v++){L=(R=17*(r[v>>1]>>4-((1&v)<<2)&15))==17*b?0:255;l[v]=L<<24|R<<16|R<<8|R}if(8==u)for(v=0;v<a;v++){L=(R=r[v])==b?0:255;l[v]=L<<24|R<<16|R<<8|R}if(16==u)for(v=0;v<a;v++){R=r[v<<1],L=d(r,v<<1)==b?0:255;l[v]=L<<24|R<<16|R<<8|R}}return s},e.decode=function(r){for(var t,n=new Uint8Array(r),i=8,a=e._bin,f=a.readUshort,o=a.readUint,s={tabs:{},frames:[]},l=new Uint8Array(n.length),c=0,u=0,d=[137,80,78,71,13,10,26,10],h=0;h<8;h++)if(n[h]!=d[h])throw"The input is not a PNG file!";for(;i<n.length;){var v=a.readUint(n,i);i+=4;var p=a.readASCII(n,i,4);if(i+=4,"IHDR"==p)e.decode._IHDR(n,i,s);else if("IDAT"==p){for(h=0;h<v;h++)l[c+h]=n[i+h];c+=v}else if("acTL"==p)s.tabs[p]={num_frames:o(n,i),num_plays:o(n,i+4)},t=new Uint8Array(n.length);else if("fcTL"==p){var b;if(0!=u)(b=s.frames[s.frames.length-1]).data=e.decode._decompress(s,t.slice(0,u),b.rect.width,b.rect.height),u=0;var g={x:o(n,i+12),y:o(n,i+16),width:o(n,i+4),height:o(n,i+8)},m=f(n,i+22);m=f(n,i+20)/(0==m?100:m);var y={rect:g,delay:Math.round(1e3*m),dispose:n[i+24],blend:n[i+25]};s.frames.push(y)}else if("fdAT"==p){for(h=0;h<v-4;h++)t[u+h]=n[i+h+4];u+=v-4}else if("pHYs"==p)s.tabs[p]=[a.readUint(n,i),a.readUint(n,i+4),n[i+8]];else if("cHRM"==p){s.tabs[p]=[];for(h=0;h<8;h++)s.tabs[p].push(a.readUint(n,i+4*h))}else if("tEXt"==p){null==s.tabs[p]&&(s.tabs[p]={});var w=a.nextZero(n,i),A=a.readASCII(n,i,w-i),U=a.readASCII(n,w+1,i+v-w-1);s.tabs[p][A]=U}else if("iTXt"==p){null==s.tabs[p]&&(s.tabs[p]={});w=0;var _=i;w=a.nextZero(n,_);A=a.readASCII(n,_,w-_),n[_=w+1],n[_+1];_+=2,w=a.nextZero(n,_);a.readASCII(n,_,w-_);_=w+1,w=a.nextZero(n,_);a.readUTF8(n,_,w-_);_=w+1;U=a.readUTF8(n,_,v-(_-i));s.tabs[p][A]=U}else if("PLTE"==p)s.tabs[p]=a.readBytes(n,i,v);else if("hIST"==p){var q=s.tabs.PLTE.length/3;s.tabs[p]=[];for(h=0;h<q;h++)s.tabs[p].push(f(n,i+2*h))}else if("tRNS"==p)3==s.ctype?s.tabs[p]=a.readBytes(n,i,v):0==s.ctype?s.tabs[p]=f(n,i):2==s.ctype&&(s.tabs[p]=[f(n,i),f(n,i+2),f(n,i+4)]);else if("gAMA"==p)s.tabs[p]=a.readUint(n,i)/1e5;else if("sRGB"==p)s.tabs[p]=n[i];else if("bKGD"==p)0==s.ctype||4==s.ctype?s.tabs[p]=[f(n,i)]:2==s.ctype||6==s.ctype?s.tabs[p]=[f(n,i),f(n,i+2),f(n,i+4)]:3==s.ctype&&(s.tabs[p]=n[i]);else if("IEND"==p)break;i+=v;a.readUint(n,i);i+=4}0!=u&&((b=s.frames[s.frames.length-1]).data=e.decode._decompress(s,t.slice(0,u),b.rect.width,b.rect.height),u=0);return s.data=e.decode._decompress(s,l,s.width,s.height),delete s.compress,delete s.interlace,delete s.filter,s},e.decode._decompress=function(r,t,n,i){return 0==r.compress&&(t=e.decode._inflate(t)),0==r.interlace?t=e.decode._filterZero(t,r,0,n,i):1==r.interlace&&(t=e.decode._readInterlace(t,r)),t},e.decode._inflate=function(e){return r.inflate(e)},e.decode._readInterlace=function(r,t){for(var n=t.width,i=t.height,a=e.decode._getBPP(t),f=a>>3,o=Math.ceil(n*a/8),s=new Uint8Array(i*o),l=0,c=[0,0,4,0,2,0,1],u=[0,4,0,2,0,1,0],d=[8,8,8,4,4,2,2],h=[8,8,4,4,2,2,1],v=0;v<7;){for(var p=d[v],b=h[v],g=0,m=0,y=c[v];y<i;)y+=p,m++;for(var w=u[v];w<n;)w+=b,g++;var A=Math.ceil(g*a/8);e.decode._filterZero(r,t,l,g,m);for(var U=0,_=c[v];_<i;){for(var q=u[v],I=l+U*A<<3;q<n;){var M;if(1==a)M=(M=r[I>>3])>>7-(7&I)&1,s[_*o+(q>>3)]|=M<<7-((3&q)<<0);if(2==a)M=(M=r[I>>3])>>6-(7&I)&3,s[_*o+(q>>2)]|=M<<6-((3&q)<<1);if(4==a)M=(M=r[I>>3])>>4-(7&I)&15,s[_*o+(q>>1)]|=M<<4-((1&q)<<2);if(a>=8)for(var T=_*o+q*f,z=0;z<f;z++)s[T+z]=r[(I>>3)+z];I+=a,q+=b}U++,_+=p}g*m!=0&&(l+=m*(1+A)),v+=1}return s},e.decode._getBPP=function(e){return[1,null,3,1,2,null,4][e.ctype]*e.depth},e.decode._filterZero=function(r,t,n,i,a){var f=e.decode._getBPP(t),o=Math.ceil(i*f/8),s=e.decode._paeth;f=Math.ceil(f/8);for(var l=0;l<a;l++){var c=n+l*o,u=c+l+1,d=r[u-1];if(0==d)for(var h=0;h<o;h++)r[c+h]=r[u+h];else if(1==d){for(h=0;h<f;h++)r[c+h]=r[u+h];for(h=f;h<o;h++)r[c+h]=r[u+h]+r[c+h-f]&255}else if(0==l){for(h=0;h<f;h++)r[c+h]=r[u+h];if(2==d)for(h=f;h<o;h++)r[c+h]=255&r[u+h];if(3==d)for(h=f;h<o;h++)r[c+h]=r[u+h]+(r[c+h-f]>>1)&255;if(4==d)for(h=f;h<o;h++)r[c+h]=r[u+h]+s(r[c+h-f],0,0)&255}else{if(2==d)for(h=0;h<o;h++)r[c+h]=r[u+h]+r[c+h-o]&255;if(3==d){for(h=0;h<f;h++)r[c+h]=r[u+h]+(r[c+h-o]>>1)&255;for(h=f;h<o;h++)r[c+h]=r[u+h]+(r[c+h-o]+r[c+h-f]>>1)&255}if(4==d){for(h=0;h<f;h++)r[c+h]=r[u+h]+s(0,r[c+h-o],0)&255;for(h=f;h<o;h++)r[c+h]=r[u+h]+s(r[c+h-f],r[c+h-o],r[c+h-f-o])&255}}}return r},e.decode._paeth=function(e,r,t){var n=e+r-t,i=Math.abs(n-e),a=Math.abs(n-r),f=Math.abs(n-t);return i<=a&&i<=f?e:a<=f?r:t},e.decode._IHDR=function(r,t,n){var i=e._bin;n.width=i.readUint(r,t),t+=4,n.height=i.readUint(r,t),t+=4,n.depth=r[t],t++,n.ctype=r[t],t++,n.compress=r[t],t++,n.filter=r[t],t++,n.interlace=r[t],t++},e._bin={nextZero:function(e,r){for(;0!=e[r];)r++;return r},readUshort:function(e,r){return e[r]<<8|e[r+1]},writeUshort:function(e,r,t){e[r]=t>>8&255,e[r+1]=255&t},readUint:function(e,r){return 16777216*e[r]+(e[r+1]<<16|e[r+2]<<8|e[r+3])},writeUint:function(e,r,t){e[r]=t>>24&255,e[r+1]=t>>16&255,e[r+2]=t>>8&255,e[r+3]=255&t},readASCII:function(e,r,t){for(var n="",i=0;i<t;i++)n+=String.fromCharCode(e[r+i]);return n},writeASCII:function(e,r,t){for(var n=0;n<t.length;n++)e[r+n]=t.charCodeAt(n)},readBytes:function(e,r,t){for(var n=[],i=0;i<t;i++)n.push(e[r+i]);return n},pad:function(e){return e.length<2?"0"+e:e},readUTF8:function(r,t,n){for(var i,a="",f=0;f<n;f++)a+="%"+e._bin.pad(r[t+f].toString(16));try{i=decodeURIComponent(a)}catch(i){return e._bin.readASCII(r,t,n)}return i}},e._copyTile=function(e,r,t,n,i,a,f,o,s){for(var l=Math.min(r,i),c=Math.min(t,a),u=0,d=0,h=0;h<c;h++)for(var v=0;v<l;v++)if(f>=0&&o>=0?(u=h*r+v<<2,d=(o+h)*i+f+v<<2):(u=(-o+h)*r-f+v<<2,d=h*i+v<<2),0==s)n[d]=e[u],n[d+1]=e[u+1],n[d+2]=e[u+2],n[d+3]=e[u+3];else if(1==s){var p=e[u+3]*(1/255),b=e[u]*p,g=e[u+1]*p,m=e[u+2]*p,y=n[d+3]*(1/255),w=n[d]*y,A=n[d+1]*y,U=n[d+2]*y,_=1-p,q=p+y*_,I=0==q?0:1/q;n[d+3]=255*q,n[d+0]=(b+w*_)*I,n[d+1]=(g+A*_)*I,n[d+2]=(m+U*_)*I}else if(2==s){p=e[u+3],b=e[u],g=e[u+1],m=e[u+2],y=n[d+3],w=n[d],A=n[d+1],U=n[d+2];p==y&&b==w&&g==A&&m==U?(n[d]=0,n[d+1]=0,n[d+2]=0,n[d+3]=0):(n[d]=b,n[d+1]=g,n[d+2]=m,n[d+3]=p)}else if(3==s){p=e[u+3],b=e[u],g=e[u+1],m=e[u+2],y=n[d+3],w=n[d],A=n[d+1],U=n[d+2];if(p==y&&b==w&&g==A&&m==U)continue;if(p<220&&y>20)return!1}return!0},e.encode=function(r,t,n,i,a,f){null==i&&(i=0),null==f&&(f=!1);var o=e.encode.compress(r,t,n,i,!1,f);return e.encode.compressPNG(o,-1),e.encode._main(o,t,n,a)},e.encodeLL=function(r,t,n,i,a,f,o){for(var s={ctype:0+(1==i?0:2)+(0==a?0:4),depth:f,frames:[]},l=(i+a)*f,c=l*t,u=0;u<r.length;u++)s.frames.push({rect:{x:0,y:0,width:t,height:n},img:new Uint8Array(r[u]),blend:0,dispose:1,bpp:Math.ceil(l/8),bpl:Math.ceil(c/8)});return e.encode.compressPNG(s,4),e.encode._main(s,t,n,o)},e.encode._main=function(r,t,n,i){var a=e.crc.crc,f=e._bin.writeUint,o=e._bin.writeUshort,s=e._bin.writeASCII,l=8,c=r.frames.length>1,u=!1,d=46+(c?20:0);if(3==r.ctype){for(var h=r.plte.length,v=0;v<h;v++)r.plte[v]>>>24!=255&&(u=!0);d+=8+3*h+4+(u?8+1*h+4:0)}for(var p=0;p<r.frames.length;p++){c&&(d+=38),d+=(q=r.frames[p]).cimg.length+12,0!=p&&(d+=4)}d+=12;var b=new Uint8Array(d),g=[137,80,78,71,13,10,26,10];for(v=0;v<8;v++)b[v]=g[v];if(f(b,l,13),s(b,l+=4,"IHDR"),f(b,l+=4,t),f(b,l+=4,n),b[l+=4]=r.depth,b[++l]=r.ctype,b[++l]=0,b[++l]=0,b[++l]=0,f(b,++l,a(b,l-17,17)),f(b,l+=4,1),s(b,l+=4,"sRGB"),b[l+=4]=1,f(b,++l,a(b,l-5,5)),l+=4,c&&(f(b,l,8),s(b,l+=4,"acTL"),f(b,l+=4,r.frames.length),f(b,l+=4,0),f(b,l+=4,a(b,l-12,12)),l+=4),3==r.ctype){f(b,l,3*(h=r.plte.length)),s(b,l+=4,"PLTE"),l+=4;for(v=0;v<h;v++){var m=3*v,y=r.plte[v],w=255&y,A=y>>>8&255,U=y>>>16&255;b[l+m+0]=w,b[l+m+1]=A,b[l+m+2]=U}if(f(b,l+=3*h,a(b,l-3*h-4,3*h+4)),l+=4,u){f(b,l,h),s(b,l+=4,"tRNS"),l+=4;for(v=0;v<h;v++)b[l+v]=r.plte[v]>>>24&255;f(b,l+=h,a(b,l-h-4,h+4)),l+=4}}var _=0;for(p=0;p<r.frames.length;p++){var q=r.frames[p];c&&(f(b,l,26),s(b,l+=4,"fcTL"),f(b,l+=4,_++),f(b,l+=4,q.rect.width),f(b,l+=4,q.rect.height),f(b,l+=4,q.rect.x),f(b,l+=4,q.rect.y),o(b,l+=4,i[p]),o(b,l+=2,1e3),b[l+=2]=q.dispose,b[++l]=q.blend,f(b,++l,a(b,l-30,30)),l+=4);var I=q.cimg;f(b,l,(h=I.length)+(0==p?0:4));var M=l+=4;s(b,l,0==p?"IDAT":"fdAT"),l+=4,0!=p&&(f(b,l,_++),l+=4);for(v=0;v<h;v++)b[l+v]=I[v];f(b,l+=h,a(b,M,l-M)),l+=4}return f(b,l,0),s(b,l+=4,"IEND"),f(b,l+=4,a(b,l-4,4)),l+=4,b.buffer},e.encode.compressPNG=function(r,t){for(var n=0;n<r.frames.length;n++){var i=r.frames[n],a=(i.rect.width,i.rect.height),f=new Uint8Array(a*i.bpl+a);i.cimg=e.encode._filterZero(i.img,a,i.bpp,i.bpl,f,t)}},e.encode.compress=function(r,t,n,i,a,f){null==f&&(f=!1);for(var o=6,s=8,l=255,c=0;c<r.length;c++)for(var u=new Uint8Array(r[c]),d=u.length,h=0;h<d;h+=4)l&=u[h+3];var v=255!=l,p=v&&a,b=e.encode.framize(r,t,n,a,p),g={},m=[],y=[];if(0!=i){var w=[];for(h=0;h<b.length;h++)w.push(b[h].img.buffer);var A=e.encode.concatRGBA(w,a),U=e.quantize(A,i),_=0,q=new Uint8Array(U.abuf);for(h=0;h<b.length;h++){var I=(F=b[h].img).length;y.push(new Uint8Array(U.inds.buffer,_>>2,I>>2));for(c=0;c<I;c+=4)F[c]=q[_+c],F[c+1]=q[_+c+1],F[c+2]=q[_+c+2],F[c+3]=q[_+c+3];_+=I}for(h=0;h<U.plte.length;h++)m.push(U.plte[h].est.rgba)}else for(c=0;c<b.length;c++){var M=b[c],T=new Uint32Array(M.img.buffer),z=M.rect.width,R=(d=T.length,new Uint8Array(d));y.push(R);for(h=0;h<d;h++){var N=T[h];if(0!=h&&N==T[h-1])R[h]=R[h-1];else if(h>z&&N==T[h-z])R[h]=R[h-z];else{var L=g[N];if(null==L&&(g[N]=L=m.length,m.push(N),m.length>=300))break;R[h]=L}}}var P=m.length;P<=256&&0==f&&(s=P<=2?1:P<=4?2:P<=16?4:8,a&&(s=8));for(c=0;c<b.length;c++){(M=b[c]).rect.x,M.rect.y,z=M.rect.width;var S=M.rect.height,D=M.img,B=(new Uint32Array(D.buffer),4*z),x=4;if(P<=256&&0==f){B=Math.ceil(s*z/8);for(var C=new Uint8Array(B*S),G=y[c],Z=0;Z<S;Z++){h=Z*B;var k=Z*z;if(8==s)for(var E=0;E<z;E++)C[h+E]=G[k+E];else if(4==s)for(E=0;E<z;E++)C[h+(E>>1)]|=G[k+E]<<4-4*(1&E);else if(2==s)for(E=0;E<z;E++)C[h+(E>>2)]|=G[k+E]<<6-2*(3&E);else if(1==s)for(E=0;E<z;E++)C[h+(E>>3)]|=G[k+E]<<7-1*(7&E)}D=C,o=3,x=1}else if(0==v&&1==b.length){C=new Uint8Array(z*S*3);var H=z*S;for(h=0;h<H;h++){var F,K=4*h;C[F=3*h]=D[K],C[F+1]=D[K+1],C[F+2]=D[K+2]}D=C,o=2,x=3,B=3*z}M.img=D,M.bpl=B,M.bpp=x}return{ctype:o,depth:s,plte:m,frames:b}},e.encode.framize=function(r,t,n,i,a){for(var f=[],o=0;o<r.length;o++){var s=new Uint8Array(r[o]),l=new Uint32Array(s.buffer),c=0,u=0,d=t,h=n,v=0;if(0==o||a)s=s.slice(0);else{for(var p=i||1==o||2==f[f.length-2].dispose?1:2,b=0,g=1e9,m=0;m<p;m++){for(var y=new Uint8Array(r[o-1-m]),w=new Uint32Array(r[o-1-m]),A=t,U=n,_=-1,q=-1,I=0;I<n;I++)for(var M=0;M<t;M++){var T=I*t+M;l[T]!=w[T]&&(M<A&&(A=M),M>_&&(_=M),I<U&&(U=I),I>q&&(q=I))}var z=-1==_?1:(_-A+1)*(q-U+1);z<g&&(g=z,b=m,-1==_?(c=u=0,d=h=1):(c=A,u=U,d=_-A+1,h=q-U+1))}y=new Uint8Array(r[o-1-b]);1==b&&(f[f.length-1].dispose=2);var R=new Uint8Array(d*h*4);new Uint32Array(R.buffer);e._copyTile(y,t,n,R,d,h,-c,-u,0),e._copyTile(s,t,n,R,d,h,-c,-u,3)?(e._copyTile(s,t,n,R,d,h,-c,-u,2),v=1):(e._copyTile(s,t,n,R,d,h,-c,-u,0),v=0),s=R}f.push({rect:{x:c,y:u,width:d,height:h},img:s,blend:v,dispose:a?1:0})}return f},e.encode._filterZero=function(t,n,i,a,f,o){if(-1!=o){for(var s=0;s<n;s++)e.encode._filterLine(f,t,s,a,i,o);return r.deflate(f)}for(var l=[],c=0;c<5;c++)if(!(n*a>5e5)||2!=c&&3!=c&&4!=c){for(s=0;s<n;s++)e.encode._filterLine(f,t,s,a,i,c);if(l.push(r.deflate(f)),1==i)break}for(var u,d=1e9,h=0;h<l.length;h++)l[h].length<d&&(u=h,d=l[h].length);return l[u]},e.encode._filterLine=function(r,t,n,i,a,f){var o=n*i,s=o+n,l=e.decode._paeth;if(r[s]=f,s++,0==f)for(var c=0;c<i;c++)r[s+c]=t[o+c];else if(1==f){for(c=0;c<a;c++)r[s+c]=t[o+c];for(c=a;c<i;c++)r[s+c]=t[o+c]-t[o+c-a]+256&255}else if(0==n){for(c=0;c<a;c++)r[s+c]=t[o+c];if(2==f)for(c=a;c<i;c++)r[s+c]=t[o+c];if(3==f)for(c=a;c<i;c++)r[s+c]=t[o+c]-(t[o+c-a]>>1)+256&255;if(4==f)for(c=a;c<i;c++)r[s+c]=t[o+c]-l(t[o+c-a],0,0)+256&255}else{if(2==f)for(c=0;c<i;c++)r[s+c]=t[o+c]+256-t[o+c-i]&255;if(3==f){for(c=0;c<a;c++)r[s+c]=t[o+c]+256-(t[o+c-i]>>1)&255;for(c=a;c<i;c++)r[s+c]=t[o+c]+256-(t[o+c-i]+t[o+c-a]>>1)&255}if(4==f){for(c=0;c<a;c++)r[s+c]=t[o+c]+256-l(0,t[o+c-i],0)&255;for(c=a;c<i;c++)r[s+c]=t[o+c]+256-l(t[o+c-a],t[o+c-i],t[o+c-a-i])&255}}},e.crc={table:function(){for(var e=new Uint32Array(256),r=0;r<256;r++){for(var t=r,n=0;n<8;n++)1&t?t=3988292384^t>>>1:t>>>=1;e[r]=t}return e}(),update:function(r,t,n,i){for(var a=0;a<i;a++)r=e.crc.table[255&(r^t[n+a])]^r>>>8;return r},crc:function(r,t,n){return 4294967295^e.crc.update(4294967295,r,t,n)}},e.quantize=function(r,t){for(var n=new Uint8Array(r),i=n.slice(0),a=new Uint32Array(i.buffer),f=e.quantize.getKDtree(i,t),o=f[0],s=f[1],l=(e.quantize.planeDst,n),c=a,u=l.length,d=new Uint8Array(n.length>>2),h=0;h<u;h+=4){var v=l[h]*(1/255),p=l[h+1]*(1/255),b=l[h+2]*(1/255),g=l[h+3]*(1/255),m=e.quantize.getNearest(o,v,p,b,g);d[h>>2]=m.ind,c[h>>2]=m.est.rgba}return{abuf:i.buffer,inds:d,plte:s}},e.quantize.getKDtree=function(r,t,n){null==n&&(n=1e-4);var i=new Uint32Array(r.buffer),a={i0:0,i1:r.length,bst:null,est:null,tdst:0,left:null,right:null};a.bst=e.quantize.stats(r,a.i0,a.i1),a.est=e.quantize.estats(a.bst);for(var f=[a];f.length<t;){for(var o=0,s=0,l=0;l<f.length;l++)f[l].est.L>o&&(o=f[l].est.L,s=l);if(o<n)break;var c=f[s],u=e.quantize.splitPixels(r,i,c.i0,c.i1,c.est.e,c.est.eMq255);if(c.i0>=u||c.i1<=u)c.est.L=0;else{var d={i0:c.i0,i1:u,bst:null,est:null,tdst:0,left:null,right:null};d.bst=e.quantize.stats(r,d.i0,d.i1),d.est=e.quantize.estats(d.bst);var h={i0:u,i1:c.i1,bst:null,est:null,tdst:0,left:null,right:null};h.bst={R:[],m:[],N:c.bst.N-d.bst.N};for(l=0;l<16;l++)h.bst.R[l]=c.bst.R[l]-d.bst.R[l];for(l=0;l<4;l++)h.bst.m[l]=c.bst.m[l]-d.bst.m[l];h.est=e.quantize.estats(h.bst),c.left=d,c.right=h,f[s]=d,f.push(h)}}f.sort(function(e,r){return r.bst.N-e.bst.N});for(l=0;l<f.length;l++)f[l].ind=l;return[a,f]},e.quantize.getNearest=function(r,t,n,i,a){if(null==r.left)return r.tdst=e.quantize.dist(r.est.q,t,n,i,a),r;var f=e.quantize.planeDst(r.est,t,n,i,a),o=r.left,s=r.right;f>0&&(o=r.right,s=r.left);var l=e.quantize.getNearest(o,t,n,i,a);if(l.tdst<=f*f)return l;var c=e.quantize.getNearest(s,t,n,i,a);return c.tdst<l.tdst?c:l},e.quantize.planeDst=function(e,r,t,n,i){var a=e.e;return a[0]*r+a[1]*t+a[2]*n+a[3]*i-e.eMq},e.quantize.dist=function(e,r,t,n,i){var a=r-e[0],f=t-e[1],o=n-e[2],s=i-e[3];return a*a+f*f+o*o+s*s},e.quantize.splitPixels=function(r,t,n,i,a,f){var o=e.quantize.vecDot;i-=4;for(;n<i;){for(;o(r,n,a)<=f;)n+=4;for(;o(r,i,a)>f;)i-=4;if(n>=i)break;var s=t[n>>2];t[n>>2]=t[i>>2],t[i>>2]=s,n+=4,i-=4}for(;o(r,n,a)>f;)n-=4;return n+4},e.quantize.vecDot=function(e,r,t){return e[r]*t[0]+e[r+1]*t[1]+e[r+2]*t[2]+e[r+3]*t[3]},e.quantize.stats=function(e,r,t){for(var n=[0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0],i=[0,0,0,0],a=t-r>>2,f=r;f<t;f+=4){var o=e[f]*(1/255),s=e[f+1]*(1/255),l=e[f+2]*(1/255),c=e[f+3]*(1/255);i[0]+=o,i[1]+=s,i[2]+=l,i[3]+=c,n[0]+=o*o,n[1]+=o*s,n[2]+=o*l,n[3]+=o*c,n[5]+=s*s,n[6]+=s*l,n[7]+=s*c,n[10]+=l*l,n[11]+=l*c,n[15]+=c*c}return n[4]=n[1],n[8]=n[2],n[9]=n[6],n[12]=n[3],n[13]=n[7],n[14]=n[11],{R:n,m:i,N:a}},e.quantize.estats=function(r){var t=r.R,n=r.m,i=r.N,a=n[0],f=n[1],o=n[2],s=n[3],l=0==i?0:1/i,c=[t[0]-a*a*l,t[1]-a*f*l,t[2]-a*o*l,t[3]-a*s*l,t[4]-f*a*l,t[5]-f*f*l,t[6]-f*o*l,t[7]-f*s*l,t[8]-o*a*l,t[9]-o*f*l,t[10]-o*o*l,t[11]-o*s*l,t[12]-s*a*l,t[13]-s*f*l,t[14]-s*o*l,t[15]-s*s*l],u=c,d=e.M4,h=[.5,.5,.5,.5],v=0,p=0;if(0!=i)for(var b=0;b<10&&(h=d.multVec(u,h),p=Math.sqrt(d.dot(h,h)),h=d.sml(1/p,h),!(Math.abs(p-v)<1e-9));b++)v=p;var g=[a*l,f*l,o*l,s*l];return{Cov:c,q:g,e:h,L:v,eMq255:d.dot(d.sml(255,g),h),eMq:d.dot(h,g),rgba:(Math.round(255*g[3])<<24|Math.round(255*g[2])<<16|Math.round(255*g[1])<<8|Math.round(255*g[0])<<0)>>>0}},e.M4={multVec:function(e,r){return[e[0]*r[0]+e[1]*r[1]+e[2]*r[2]+e[3]*r[3],e[4]*r[0]+e[5]*r[1]+e[6]*r[2]+e[7]*r[3],e[8]*r[0]+e[9]*r[1]+e[10]*r[2]+e[11]*r[3],e[12]*r[0]+e[13]*r[1]+e[14]*r[2]+e[15]*r[3]]},dot:function(e,r){return e[0]*r[0]+e[1]*r[1]+e[2]*r[2]+e[3]*r[3]},sml:function(e,r){return[e*r[0],e*r[1],e*r[2],e*r[3]]}},e.encode.concatRGBA=function(e,r){for(var t=0,n=0;n<e.length;n++)t+=e[n].byteLength;var i=new Uint8Array(t),a=0;for(n=0;n<e.length;n++){for(var f=new Uint8Array(e[n]),o=f.length,s=0;s<o;s+=4){var l=f[s],c=f[s+1],u=f[s+2],d=f[s+3];r&&(d=0==(128&d)?0:255),0==d&&(l=c=u=0),i[a+s]=l,i[a+s+1]=c,i[a+s+2]=u,i[a+s+3]=d}a+=o}return i.buffer}}(e,"function"==typeof require?require("pako"):window.pako)}();
+</script>
+
+<script>
+    class Player {
+
+        constructor(container) {
+            this.container = container
+            this.global_frac = 0.0
+            this.container = document.getElementById(container)
+            this.progress = null;
+            this.mat = [[]]
+
+            this.player = this.container.querySelector('audio')
+            this.demo_img = this.container.querySelector('.underlay > img')
+            this.overlay = this.container.querySelector('.overlay')
+            this.playpause = this.container.querySelector(".playpause");
+            this.download = this.container.querySelector(".download");
+            this.play_img = this.container.querySelector('.play-img')
+            this.pause_img = this.container.querySelector('.pause-img')
+            this.canvas = this.container.querySelector('.response-canvas')
+            this.response_container = this.container.querySelector('.response')
+            this.context = this.canvas.getContext('2d');
+
+            // console.log(this.player.duration)
+            var togglePlayPause = () => {
+                if (this.player.networkState !== 1) {
+                    return
+                }
+                if (this.player.paused || this.player.ended) {
+                    this.play()
+                } else {
+                    this.pause()
+                }
+            }
+
+            this.update = () => {
+                this.global_frac = this.player.currentTime / this.player.duration
+                // this.global_frac = frac
+                // console.log(this.player.currentTime, this.player.duration, this.global_frac)
+                this.overlay.style.width = (100*(1.0 - this.global_frac)).toString() + '%'
+                this.redraw()
+            }
+
+            // var start = null;
+            this.updateLoop = (timestamp) => {
+                // if (!start) start = timestamp;
+                // var progress = timestamp - start;
+                this.update()
+                // this.progress = setTimeout(this.updateLoop, 10)
+                this.progress = window.requestAnimationFrame(this.updateLoop)
+            }
+
+            this.seek = (e) => {
+                this.global_frac = e.offsetX / this.demo_img.width
+                this.player.currentTime = this.global_frac * this.player.duration
+                // console.log(this.global_frac)
+                this.overlay.style.width = (100*(1.0 - this.global_frac)).toString() + '%'
+                this.redraw()
+            }
+
+            var download_audio = () => {
+                var url = this.player.querySelector('#src').src
+                const a = document.createElement('a')
+                a.href = url
+                a.download = "download"
+                document.body.appendChild(a)
+                a.click()
+                document.body.removeChild(a)
+            }
+
+            this.demo_img.onclick = this.seek;
+            this.playpause.disabled = true
+            this.player.onplay = this.updateLoop
+            this.player.onpause = () => {
+                window.cancelAnimationFrame(this.progress)
+                this.update();
+            }
+            this.player.onended = () => {this.pause()}
+            this.playpause.onclick = togglePlayPause;
+            this.download.onclick = download_audio;
+        }
+
+        load(audio_fname, img_fname, levels_fname) {
+            this.pause()
+            window.cancelAnimationFrame(this.progress)
+            this.playpause.disabled = true
+
+            this.player.querySelector('#src').setAttribute("src", audio_fname)
+            this.player.load()
+            this.demo_img.setAttribute("src", img_fname)
+            this.overlay.style.width = '0%'
+
+            fetch(levels_fname)
+            .then(response => response.arrayBuffer())
+            .then(text => {
+                this.mat = this.parse(text);
+                this.playpause.disabled = false;
+                this.redraw();
+            })
+        }
+
+        parse(buffer) {
+            var img = UPNG.decode(buffer)
+            var dat = UPNG.toRGBA8(img)[0]
+            var view = new DataView(dat)
+            var data = new Array(img.width).fill(0).map(() => new Array(img.height).fill(0));
+
+            var min =100
+            var max = -100
+            var idx = 0
+            for (let i=0; i < img.height*img.width*4; i+=4) {
+                var rgba = [view.getUint8(i, 1) / 255, view.getUint8(i + 1, 1) / 255, view.getUint8(i + 2, 1) / 255, view.getUint8(i + 3, 1) / 255]
+                var norm = Math.pow(Math.pow(rgba[0], 2) + Math.pow(rgba[1], 2) + Math.pow(rgba[2], 2), 0.5)
+                data[idx % img.width][img.height - Math.floor(idx / img.width) - 1] = norm
+
+                idx += 1
+                min = Math.min(min, norm)
+                max = Math.max(max, norm)
+            }
+            for (let i = 0; i < data.length; i++) {
+                for (let j = 0; j < data[i].length; j++) {
+                    data[i][j] = Math.pow((data[i][j] - min) / (max - min), 1.5)
+                }
+            }
+            var data3 = new Array(img.width).fill(0).map(() => new Array(img.height).fill(0));
+            for (let i = 0; i < data.length; i++) {
+                for (let j = 0; j < data[i].length; j++) {
+                    if (i == 0 || i == (data.length - 1)) {
+                        data3[i][j] = data[i][j]
+                    } else{
+                        data3[i][j] = 0.33*(data[i - 1][j]) + 0.33*(data[i][j]) + 0.33*(data[i + 1][j])
+                        // data3[i][j] = 0.00*(data[i - 1][j]) + 1.00*(data[i][j]) + 0.00*(data[i + 1][j])
+                    }
+                }
+            }
+
+            var scale = 5
+            var data2 = new Array(scale*img.width).fill(0).map(() => new Array(img.height).fill(0));
+            for (let j = 0; j < data[0].length; j++) {
+                for (let i = 0; i < data.length - 1; i++) {
+                    for (let k = 0; k < scale; k++) {
+                        data2[scale*i + k][j] = (1.0 - (k/scale))*data3[i][j] + (k / scale)*data3[i + 1][j]
+                    }
+                }
+            }
+            return data2
+        }
+
+        play() {
+            this.player.play();
+            this.play_img.style.display = 'none'
+            this.pause_img.style.display = 'block'
+        }
+
+        pause() {
+            this.player.pause();
+            this.pause_img.style.display = 'none'
+            this.play_img.style.display = 'block'
+        }
+
+        redraw() {
+            this.canvas.width = window.devicePixelRatio*this.response_container.offsetWidth;
+            this.canvas.height = window.devicePixelRatio*this.response_container.offsetHeight;
+
+            this.context.clearRect(0, 0, this.canvas.width, this.canvas.height)
+            this.canvas.style.width = (this.canvas.width / window.devicePixelRatio).toString() + "px";
+            this.canvas.style.height = (this.canvas.height / window.devicePixelRatio).toString() + "px";
+
+            var f = this.global_frac*this.mat.length
+            var tstep = Math.min(Math.floor(f), this.mat.length - 2)
+            var heights = this.mat[tstep]
+            var bar_width = (this.canvas.width / heights.length) - 1
+
+            for (let k = 0; k < heights.length - 1; k++) {
+                var height = Math.max(Math.round((heights[k])*this.canvas.height), 3)
+                this.context.fillStyle = '#696f7b';
+                this.context.fillRect(k*(bar_width + 1), (this.canvas.height - height) / 2, bar_width, height);
+            }
+        }
+    }
+</script>

dac-vae/audiotools/core/templates/pandoc.css

@@ -0,0 +1,407 @@
+/*
+Copyright (c) 2017 Chris Patuzzo
+https://twitter.com/chrispatuzzo
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
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dac-vae/audiotools/core/templates/widget.html

@@ -0,0 +1,52 @@
+<div id='PLAYER_ID' class='player' style="max-width: MAX_WIDTH;">
+    <div class='spectrogram' style="padding-top: PADDING_AMOUNT;">
+        <div class='overlay'></div>
+        <div class='underlay'>
+            <img>
+        </div>
+    </div>
+
+    <div class='audio-controls'>
+        <button id="playpause" disabled class='playpause' title="play">
+            <svg class='play-img' width="14px" height="19px" viewBox="0 0 14 19">
+                <polygon id="Triangle" fill="#000000" transform="translate(9, 9.5) rotate(90) translate(-7, -9.5) " points="7 2.5 16.5 16.5 -2.5 16.5"></polygon>
+            </svg>
+            <svg class='pause-img' width="16px" height="19px" viewBox="0 0 16 19">
+                <g fill="#000000" stroke="#000000">
+                    <rect id="Rectangle" x="0.5" y="0.5" width="4" height="18"></rect>
+                    <rect id="Rectangle" x="11.5" y="0.5" width="4" height="18"></rect>
+                </g>
+            </svg>
+        </button>
+
+        <audio class='play'>
+            <source id='src'>
+        </audio>
+        <div class='response'>
+            <canvas class='response-canvas'></canvas>
+        </div>
+
+        <button id="download" class='download' title="download">
+            <svg class='download-img' x="0px" y="0px" viewBox="0 0 29.978 29.978" style="enable-background:new 0 0 29.978 29.978;" xml:space="preserve">
+            <g>
+                <path d="M25.462,19.105v6.848H4.515v-6.848H0.489v8.861c0,1.111,0.9,2.012,2.016,2.012h24.967c1.115,0,2.016-0.9,2.016-2.012
+                    v-8.861H25.462z"/>
+                <path d="M14.62,18.426l-5.764-6.965c0,0-0.877-0.828,0.074-0.828s3.248,0,3.248,0s0-0.557,0-1.416c0-2.449,0-6.906,0-8.723
+                    c0,0-0.129-0.494,0.615-0.494c0.75,0,4.035,0,4.572,0c0.536,0,0.524,0.416,0.524,0.416c0,1.762,0,6.373,0,8.742
+                    c0,0.768,0,1.266,0,1.266s1.842,0,2.998,0c1.154,0,0.285,0.867,0.285,0.867s-4.904,6.51-5.588,7.193
+                    C15.092,18.979,14.62,18.426,14.62,18.426z"/>
+            </g>
+            </svg>
+        </button>
+    </div>
+</div>
+
+<script>
+    var PLAYER_ID = new Player('PLAYER_ID')
+    PLAYER_ID.load(
+        "AUDIO_SRC",
+        "IMAGE_SRC",
+        "LEVELS_SRC"
+    )
+    window.addEventListener("resize", function() {PLAYER_ID.redraw()})
+</script>

dac-vae/audiotools/core/util.py

@@ -0,0 +1,671 @@
+import csv
+import glob
+import math
+import numbers
+import os
+import random
+import typing
+from contextlib import contextmanager
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Dict
+from typing import List
+
+import numpy as np
+import torch
+import torchaudio
+from flatten_dict import flatten
+from flatten_dict import unflatten
+
+
+@dataclass
+class Info:
+    """Shim for torchaudio.info API changes."""
+
+    sample_rate: float
+    num_frames: int
+
+    @property
+    def duration(self) -> float:
+        return self.num_frames / self.sample_rate
+
+
+def info(audio_path: str):
+    """Shim for torchaudio.info to make 0.7.2 API match 0.8.0.
+
+    Parameters
+    ----------
+    audio_path : str
+        Path to audio file.
+    """
+    # try default backend first, then fallback to soundfile
+    try:
+        info = torchaudio.info(str(audio_path))
+    except:  # pragma: no cover
+        info = torchaudio.backend.soundfile_backend.info(str(audio_path))
+
+    if isinstance(info, tuple):  # pragma: no cover
+        signal_info = info[0]
+        info = Info(sample_rate=signal_info.rate, num_frames=signal_info.length)
+    else:
+        info = Info(sample_rate=info.sample_rate, num_frames=info.num_frames)
+
+    return info
+
+
+def ensure_tensor(
+    x: typing.Union[np.ndarray, torch.Tensor, float, int],
+    ndim: int = None,
+    batch_size: int = None,
+):
+    """Ensures that the input ``x`` is a tensor of specified
+    dimensions and batch size.
+
+    Parameters
+    ----------
+    x : typing.Union[np.ndarray, torch.Tensor, float, int]
+        Data that will become a tensor on its way out.
+    ndim : int, optional
+        How many dimensions should be in the output, by default None
+    batch_size : int, optional
+        The batch size of the output, by default None
+
+    Returns
+    -------
+    torch.Tensor
+        Modified version of ``x`` as a tensor.
+    """
+    if not torch.is_tensor(x):
+        x = torch.as_tensor(x)
+    if ndim is not None:
+        assert x.ndim <= ndim
+        while x.ndim < ndim:
+            x = x.unsqueeze(-1)
+    if batch_size is not None:
+        if x.shape[0] != batch_size:
+            shape = list(x.shape)
+            shape[0] = batch_size
+            x = x.expand(*shape)
+    return x
+
+
+def _get_value(other):
+    from . import AudioSignal
+
+    if isinstance(other, AudioSignal):
+        return other.audio_data
+    return other
+
+
+def hz_to_bin(hz: torch.Tensor, n_fft: int, sample_rate: int):
+    """Closest frequency bin given a frequency, number
+    of bins, and a sampling rate.
+
+    Parameters
+    ----------
+    hz : torch.Tensor
+       Tensor of frequencies in Hz.
+    n_fft : int
+        Number of FFT bins.
+    sample_rate : int
+        Sample rate of audio.
+
+    Returns
+    -------
+    torch.Tensor
+        Closest bins to the data.
+    """
+    shape = hz.shape
+    hz = hz.flatten()
+    freqs = torch.linspace(0, sample_rate / 2, 2 + n_fft // 2)
+    hz[hz > sample_rate / 2] = sample_rate / 2
+
+    closest = (hz[None, :] - freqs[:, None]).abs()
+    closest_bins = closest.min(dim=0).indices
+
+    return closest_bins.reshape(*shape)
+
+
+def random_state(seed: typing.Union[int, np.random.RandomState]):
+    """
+    Turn seed into a np.random.RandomState instance.
+
+    Parameters
+    ----------
+    seed : typing.Union[int, np.random.RandomState] or None
+        If seed is None, return the RandomState singleton used by np.random.
+        If seed is an int, return a new RandomState instance seeded with seed.
+        If seed is already a RandomState instance, return it.
+        Otherwise raise ValueError.
+
+    Returns
+    -------
+    np.random.RandomState
+        Random state object.
+
+    Raises
+    ------
+    ValueError
+        If seed is not valid, an error is thrown.
+    """
+    if seed is None or seed is np.random:
+        return np.random.mtrand._rand
+    elif isinstance(seed, (numbers.Integral, np.integer, int)):
+        return np.random.RandomState(seed)
+    elif isinstance(seed, np.random.RandomState):
+        return seed
+    else:
+        raise ValueError(
+            "%r cannot be used to seed a numpy.random.RandomState" " instance" % seed
+        )
+
+
+def seed(random_seed, set_cudnn=False):
+    """
+    Seeds all random states with the same random seed
+    for reproducibility. Seeds ``numpy``, ``random`` and ``torch``
+    random generators.
+    For full reproducibility, two further options must be set
+    according to the torch documentation:
+    https://pytorch.org/docs/stable/notes/randomness.html
+    To do this, ``set_cudnn`` must be True. It defaults to
+    False, since setting it to True results in a performance
+    hit.
+
+    Args:
+        random_seed (int): integer corresponding to random seed to
+        use.
+        set_cudnn (bool): Whether or not to set cudnn into determinstic
+        mode and off of benchmark mode. Defaults to False.
+    """
+
+    torch.manual_seed(random_seed)
+    np.random.seed(random_seed)
+    random.seed(random_seed)
+
+    if set_cudnn:
+        torch.backends.cudnn.deterministic = True
+        torch.backends.cudnn.benchmark = False
+
+
+@contextmanager
+def _close_temp_files(tmpfiles: list):
+    """Utility function for creating a context and closing all temporary files
+    once the context is exited. For correct functionality, all temporary file
+    handles created inside the context must be appended to the ```tmpfiles```
+    list.
+
+    This function is taken wholesale from Scaper.
+
+    Parameters
+    ----------
+    tmpfiles : list
+        List of temporary file handles
+    """
+
+    def _close():
+        for t in tmpfiles:
+            try:
+                t.close()
+                os.unlink(t.name)
+            except:
+                pass
+
+    try:
+        yield
+    except:  # pragma: no cover
+        _close()
+        raise
+    _close()
+
+
+AUDIO_EXTENSIONS = [".wav", ".flac", ".mp3", ".mp4"]
+
+
+def find_audio(folder: str, ext: List[str] = AUDIO_EXTENSIONS):
+    """Finds all audio files in a directory recursively.
+    Returns a list.
+
+    Parameters
+    ----------
+    folder : str
+        Folder to look for audio files in, recursively.
+    ext : List[str], optional
+        Extensions to look for without the ., by default
+        ``['.wav', '.flac', '.mp3', '.mp4']``.
+    """
+    folder = Path(folder)
+    # Take care of case where user has passed in an audio file directly
+    # into one of the calling functions.
+    if str(folder).endswith(tuple(ext)):
+        # if, however, there's a glob in the path, we need to
+        # return the glob, not the file.
+        if "*" in str(folder):
+            return glob.glob(str(folder), recursive=("**" in str(folder)))
+        else:
+            return [folder]
+
+    files = []
+    for x in ext:
+        files += folder.glob(f"**/*{x}")
+    return files
+
+
+def read_sources(
+    sources: List[str],
+    remove_empty: bool = True,
+    relative_path: str = "",
+    ext: List[str] = AUDIO_EXTENSIONS,
+):
+    """Reads audio sources that can either be folders
+    full of audio files, or CSV files that contain paths
+    to audio files. CSV files that adhere to the expected
+    format can be generated by
+    :py:func:`audiotools.data.preprocess.create_csv`.
+
+    Parameters
+    ----------
+    sources : List[str]
+        List of audio sources to be converted into a
+        list of lists of audio files.
+    remove_empty : bool, optional
+        Whether or not to remove rows with an empty "path"
+        from each CSV file, by default True.
+
+    Returns
+    -------
+    list
+        List of lists of rows of CSV files.
+    """
+    files = []
+    relative_path = Path(relative_path)
+    for source in sources:
+        source = str(source)
+        _files = []
+        if source.endswith(".csv"):
+            with open(source, "r") as f:
+                reader = csv.DictReader(f)
+                for x in reader:
+                    if remove_empty and x["path"] == "":
+                        continue
+                    if x["path"] != "":
+                        x["path"] = str(relative_path / x["path"])
+                    _files.append(x)
+        else:
+            for x in find_audio(source, ext=ext):
+                x = str(relative_path / x)
+                _files.append({"path": x})
+        files.append(sorted(_files, key=lambda x: x["path"]))
+    return files
+
+
+def choose_from_list_of_lists(
+    state: np.random.RandomState, list_of_lists: list, p: float = None
+):
+    """Choose a single item from a list of lists.
+
+    Parameters
+    ----------
+    state : np.random.RandomState
+        Random state to use when choosing an item.
+    list_of_lists : list
+        A list of lists from which items will be drawn.
+    p : float, optional
+        Probabilities of each list, by default None
+
+    Returns
+    -------
+    typing.Any
+        An item from the list of lists.
+    """
+    source_idx = state.choice(list(range(len(list_of_lists))), p=p)
+    item_idx = state.randint(len(list_of_lists[source_idx]))
+    return list_of_lists[source_idx][item_idx], source_idx, item_idx
+
+
+@contextmanager
+def chdir(newdir: typing.Union[Path, str]):
+    """
+    Context manager for switching directories to run a
+    function. Useful for when you want to use relative
+    paths to different runs.
+
+    Parameters
+    ----------
+    newdir : typing.Union[Path, str]
+        Directory to switch to.
+    """
+    curdir = os.getcwd()
+    try:
+        os.chdir(newdir)
+        yield
+    finally:
+        os.chdir(curdir)
+
+
+def prepare_batch(batch: typing.Union[dict, list, torch.Tensor], device: str = "cpu"):
+    """Moves items in a batch (typically generated by a DataLoader as a list
+    or a dict) to the specified device. This works even if dictionaries
+    are nested.
+
+    Parameters
+    ----------
+    batch : typing.Union[dict, list, torch.Tensor]
+        Batch, typically generated by a dataloader, that will be moved to
+        the device.
+    device : str, optional
+        Device to move batch to, by default "cpu"
+
+    Returns
+    -------
+    typing.Union[dict, list, torch.Tensor]
+        Batch with all values moved to the specified device.
+    """
+    if isinstance(batch, dict):
+        batch = flatten(batch)
+        for key, val in batch.items():
+            try:
+                batch[key] = val.to(device)
+            except:
+                pass
+        batch = unflatten(batch)
+    elif torch.is_tensor(batch):
+        batch = batch.to(device)
+    elif isinstance(batch, list):
+        for i in range(len(batch)):
+            try:
+                batch[i] = batch[i].to(device)
+            except:
+                pass
+    return batch
+
+
+def sample_from_dist(dist_tuple: tuple, state: np.random.RandomState = None):
+    """Samples from a distribution defined by a tuple. The first
+    item in the tuple is the distribution type, and the rest of the
+    items are arguments to that distribution. The distribution function
+    is gotten from the ``np.random.RandomState`` object.
+
+    Parameters
+    ----------
+    dist_tuple : tuple
+        Distribution tuple
+    state : np.random.RandomState, optional
+        Random state, or seed to use, by default None
+
+    Returns
+    -------
+    typing.Union[float, int, str]
+        Draw from the distribution.
+
+    Examples
+    --------
+    Sample from a uniform distribution:
+
+    >>> dist_tuple = ("uniform", 0, 1)
+    >>> sample_from_dist(dist_tuple)
+
+    Sample from a constant distribution:
+
+    >>> dist_tuple = ("const", 0)
+    >>> sample_from_dist(dist_tuple)
+
+    Sample from a normal distribution:
+
+    >>> dist_tuple = ("normal", 0, 0.5)
+    >>> sample_from_dist(dist_tuple)
+
+    """
+    if dist_tuple[0] == "const":
+        return dist_tuple[1]
+    state = random_state(state)
+    dist_fn = getattr(state, dist_tuple[0])
+    return dist_fn(*dist_tuple[1:])
+
+
+def collate(list_of_dicts: list, n_splits: int = None):
+    """Collates a list of dictionaries (e.g. as returned by a
+    dataloader) into a dictionary with batched values. This routine
+    uses the default torch collate function for everything
+    except AudioSignal objects, which are handled by the
+    :py:func:`audiotools.core.audio_signal.AudioSignal.batch`
+    function.
+
+    This function takes n_splits to enable splitting a batch
+    into multiple sub-batches for the purposes of gradient accumulation,
+    etc.
+
+    Parameters
+    ----------
+    list_of_dicts : list
+        List of dictionaries to be collated.
+    n_splits : int
+        Number of splits to make when creating the batches (split into
+        sub-batches). Useful for things like gradient accumulation.
+
+    Returns
+    -------
+    dict
+        Dictionary containing batched data.
+    """
+
+    from . import AudioSignal
+
+    batches = []
+    list_len = len(list_of_dicts)
+
+    return_list = False if n_splits is None else True
+    n_splits = 1 if n_splits is None else n_splits
+    n_items = int(math.ceil(list_len / n_splits))
+
+    for i in range(0, list_len, n_items):
+        # Flatten the dictionaries to avoid recursion.
+        list_of_dicts_ = [flatten(d) for d in list_of_dicts[i : i + n_items]]
+        dict_of_lists = {
+            k: [dic[k] for dic in list_of_dicts_] for k in list_of_dicts_[0]
+        }
+
+        batch = {}
+        for k, v in dict_of_lists.items():
+            if isinstance(v, list):
+                if all(isinstance(s, AudioSignal) for s in v):
+                    batch[k] = AudioSignal.batch(v, pad_signals=True)
+                else:
+                    # Borrow the default collate fn from torch.
+                    batch[k] = torch.utils.data._utils.collate.default_collate(v)
+        batches.append(unflatten(batch))
+
+    batches = batches[0] if not return_list else batches
+    return batches
+
+
+BASE_SIZE = 864
+DEFAULT_FIG_SIZE = (9, 3)
+
+
+def format_figure(
+    fig_size: tuple = None,
+    title: str = None,
+    fig=None,
+    format_axes: bool = True,
+    format: bool = True,
+    font_color: str = "white",
+):
+    """Prettifies the spectrogram and waveform plots. A title
+    can be inset into the top right corner, and the axes can be
+    inset into the figure, allowing the data to take up the entire
+    image. Used in
+
+    - :py:func:`audiotools.core.display.DisplayMixin.specshow`
+    - :py:func:`audiotools.core.display.DisplayMixin.waveplot`
+    - :py:func:`audiotools.core.display.DisplayMixin.wavespec`
+
+    Parameters
+    ----------
+    fig_size : tuple, optional
+        Size of figure, by default (9, 3)
+    title : str, optional
+        Title to inset in top right, by default None
+    fig : matplotlib.figure.Figure, optional
+        Figure object, if None ``plt.gcf()`` will be used, by default None
+    format_axes : bool, optional
+        Format the axes to be inside the figure, by default True
+    format : bool, optional
+        This formatting can be skipped entirely by passing ``format=False``
+        to any of the plotting functions that use this formater, by default True
+    font_color : str, optional
+        Color of font of axes, by default "white"
+    """
+    import matplotlib
+    import matplotlib.pyplot as plt
+
+    if fig_size is None:
+        fig_size = DEFAULT_FIG_SIZE
+    if not format:
+        return
+    if fig is None:
+        fig = plt.gcf()
+    fig.set_size_inches(*fig_size)
+    axs = fig.axes
+
+    pixels = (fig.get_size_inches() * fig.dpi)[0]
+    font_scale = pixels / BASE_SIZE
+
+    if format_axes:
+        axs = fig.axes
+
+        for ax in axs:
+            ymin, _ = ax.get_ylim()
+            xmin, _ = ax.get_xlim()
+
+            ticks = ax.get_yticks()
+            for t in ticks[2:-1]:
+                t = axs[0].annotate(
+                    f"{(t / 1000):2.1f}k",
+                    xy=(xmin, t),
+                    xycoords="data",
+                    xytext=(5, -5),
+                    textcoords="offset points",
+                    ha="left",
+                    va="top",
+                    color=font_color,
+                    fontsize=12 * font_scale,
+                    alpha=0.75,
+                )
+
+            ticks = ax.get_xticks()[2:]
+            for t in ticks[:-1]:
+                t = axs[0].annotate(
+                    f"{t:2.1f}s",
+                    xy=(t, ymin),
+                    xycoords="data",
+                    xytext=(5, 5),
+                    textcoords="offset points",
+                    ha="center",
+                    va="bottom",
+                    color=font_color,
+                    fontsize=12 * font_scale,
+                    alpha=0.75,
+                )
+
+            ax.margins(0, 0)
+            ax.set_axis_off()
+            ax.xaxis.set_major_locator(plt.NullLocator())
+            ax.yaxis.set_major_locator(plt.NullLocator())
+
+        plt.subplots_adjust(top=1, bottom=0, right=1, left=0, hspace=0, wspace=0)
+
+    if title is not None:
+        t = axs[0].annotate(
+            title,
+            xy=(1, 1),
+            xycoords="axes fraction",
+            fontsize=20 * font_scale,
+            xytext=(-5, -5),
+            textcoords="offset points",
+            ha="right",
+            va="top",
+            color="white",
+        )
+        t.set_bbox(dict(facecolor="black", alpha=0.5, edgecolor="black"))
+
+
+def generate_chord_dataset(
+    max_voices: int = 8,
+    sample_rate: int = 44100,
+    num_items: int = 5,
+    duration: float = 1.0,
+    min_note: str = "C2",
+    max_note: str = "C6",
+    output_dir: Path = "chords",
+):
+    """
+    Generates a toy multitrack dataset of chords, synthesized from sine waves.
+
+
+    Parameters
+    ----------
+    max_voices : int, optional
+        Maximum number of voices in a chord, by default 8
+    sample_rate : int, optional
+        Sample rate of audio, by default 44100
+    num_items : int, optional
+        Number of items to generate, by default 5
+    duration : float, optional
+        Duration of each item, by default 1.0
+    min_note : str, optional
+        Minimum note in the dataset, by default "C2"
+    max_note : str, optional
+        Maximum note in the dataset, by default "C6"
+    output_dir : Path, optional
+        Directory to save the dataset, by default "chords"
+
+    """
+    import librosa
+    from . import AudioSignal
+    from ..data.preprocess import create_csv
+
+    min_midi = librosa.note_to_midi(min_note)
+    max_midi = librosa.note_to_midi(max_note)
+
+    tracks = []
+    for idx in range(num_items):
+        track = {}
+        # figure out how many voices to put in this track
+        num_voices = random.randint(1, max_voices)
+        for voice_idx in range(num_voices):
+            # choose some random params
+            midinote = random.randint(min_midi, max_midi)
+            dur = random.uniform(0.85 * duration, duration)
+
+            sig = AudioSignal.wave(
+                frequency=librosa.midi_to_hz(midinote),
+                duration=dur,
+                sample_rate=sample_rate,
+                shape="sine",
+            )
+            track[f"voice_{voice_idx}"] = sig
+        tracks.append(track)
+
+    # save the tracks to disk
+    output_dir = Path(output_dir)
+    output_dir.mkdir(exist_ok=True)
+    for idx, track in enumerate(tracks):
+        track_dir = output_dir / f"track_{idx}"
+        track_dir.mkdir(exist_ok=True)
+        for voice_name, sig in track.items():
+            sig.write(track_dir / f"{voice_name}.wav")
+
+    all_voices = list(set([k for track in tracks for k in track.keys()]))
+    voice_lists = {voice: [] for voice in all_voices}
+    for track in tracks:
+        for voice_name in all_voices:
+            if voice_name in track:
+                voice_lists[voice_name].append(track[voice_name].path_to_file)
+            else:
+                voice_lists[voice_name].append("")
+
+    for voice_name, paths in voice_lists.items():
+        create_csv(paths, output_dir / f"{voice_name}.csv", loudness=True)
+
+    return output_dir

dac-vae/audiotools/core/whisper.py

@@ -0,0 +1,97 @@
+import torch
+
+
+class WhisperMixin:
+    is_initialized = False
+
+    def setup_whisper(
+        self,
+        pretrained_model_name_or_path: str = "openai/whisper-base.en",
+        device: str = torch.device("cuda" if torch.cuda.is_available() else "cpu"),
+    ):
+        from transformers import WhisperForConditionalGeneration
+        from transformers import WhisperProcessor
+
+        self.whisper_device = device
+        self.whisper_processor = WhisperProcessor.from_pretrained(
+            pretrained_model_name_or_path
+        )
+        self.whisper_model = WhisperForConditionalGeneration.from_pretrained(
+            pretrained_model_name_or_path
+        ).to(self.whisper_device)
+        self.is_initialized = True
+
+    def get_whisper_features(self) -> torch.Tensor:
+        """Preprocess audio signal as per the whisper model's training config.
+
+        Returns
+        -------
+        torch.Tensor
+            The prepinput features of the audio signal. Shape: (1, channels, seq_len)
+        """
+        import torch
+
+        if not self.is_initialized:
+            self.setup_whisper()
+
+        signal = self.to(self.device)
+        raw_speech = list(
+            (
+                signal.clone()
+                .resample(self.whisper_processor.feature_extractor.sampling_rate)
+                .audio_data[:, 0, :]
+                .numpy()
+            )
+        )
+
+        with torch.inference_mode():
+            input_features = self.whisper_processor(
+                raw_speech,
+                sampling_rate=self.whisper_processor.feature_extractor.sampling_rate,
+                return_tensors="pt",
+            ).input_features
+
+        return input_features
+
+    def get_whisper_transcript(self) -> str:
+        """Get the transcript of the audio signal using the whisper model.
+
+        Returns
+        -------
+        str
+            The transcript of the audio signal, including special tokens such as <|startoftranscript|> and <|endoftext|>.
+        """
+
+        if not self.is_initialized:
+            self.setup_whisper()
+
+        input_features = self.get_whisper_features()
+
+        with torch.inference_mode():
+            input_features = input_features.to(self.whisper_device)
+            generated_ids = self.whisper_model.generate(inputs=input_features)
+
+        transcription = self.whisper_processor.batch_decode(generated_ids)
+        return transcription[0]
+
+    def get_whisper_embeddings(self) -> torch.Tensor:
+        """Get the last hidden state embeddings of the audio signal using the whisper model.
+
+        Returns
+        -------
+        torch.Tensor
+            The Whisper embeddings of the audio signal. Shape: (1, seq_len, hidden_size)
+        """
+        import torch
+
+        if not self.is_initialized:
+            self.setup_whisper()
+
+        input_features = self.get_whisper_features()
+        encoder = self.whisper_model.get_encoder()
+
+        with torch.inference_mode():
+            input_features = input_features.to(self.whisper_device)
+            embeddings = encoder(input_features)
+
+        return embeddings.last_hidden_state

dac-vae/audiotools/data/__init__.py

@@ -0,0 +1,3 @@
+from . import datasets
+from . import preprocess
+from . import transforms

dac-vae/audiotools/data/datasets.py

@@ -0,0 +1,517 @@
+from pathlib import Path
+from typing import Callable
+from typing import Dict
+from typing import List
+from typing import Union
+
+import numpy as np
+from torch.utils.data import SequentialSampler
+from torch.utils.data.distributed import DistributedSampler
+
+from ..core import AudioSignal
+from ..core import util
+
+
+class AudioLoader:
+    """Loads audio endlessly from a list of audio sources
+    containing paths to audio files. Audio sources can be
+    folders full of audio files (which are found via file
+    extension) or by providing a CSV file which contains paths
+    to audio files.
+
+    Parameters
+    ----------
+    sources : List[str], optional
+        Sources containing folders, or CSVs with
+        paths to audio files, by default None
+    weights : List[float], optional
+        Weights to sample audio files from each source, by default None
+    relative_path : str, optional
+        Path audio should be loaded relative to, by default ""
+    transform : Callable, optional
+        Transform to instantiate alongside audio sample,
+        by default None
+    ext : List[str]
+        List of extensions to find audio within each source by. Can
+        also be a file name (e.g. "vocals.wav"). by default
+        ``['.wav', '.flac', '.mp3', '.mp4']``.
+    shuffle: bool
+        Whether to shuffle the files within the dataloader. Defaults to True.
+    shuffle_state: int
+        State to use to seed the shuffle of the files.
+    """
+
+    def __init__(
+        self,
+        sources: List[str] = None,
+        weights: List[float] = None,
+        transform: Callable = None,
+        relative_path: str = "",
+        ext: List[str] = util.AUDIO_EXTENSIONS,
+        shuffle: bool = True,
+        shuffle_state: int = 0,
+    ):
+        self.audio_lists = util.read_sources(
+            sources, relative_path=relative_path, ext=ext
+        )
+
+        self.audio_indices = [
+            (src_idx, item_idx)
+            for src_idx, src in enumerate(self.audio_lists)
+            for item_idx in range(len(src))
+        ]
+        if shuffle:
+            state = util.random_state(shuffle_state)
+            state.shuffle(self.audio_indices)
+
+        self.sources = sources
+        self.weights = weights
+        self.transform = transform
+
+    def __call__(
+        self,
+        state,
+        sample_rate: int,
+        duration: float,
+        loudness_cutoff: float = -40,
+        num_channels: int = 1,
+        offset: float = None,
+        source_idx: int = None,
+        item_idx: int = None,
+        global_idx: int = None,
+    ):
+        if source_idx is not None and item_idx is not None:
+            try:
+                audio_info = self.audio_lists[source_idx][item_idx]
+            except:
+                audio_info = {"path": "none"}
+        elif global_idx is not None:
+            source_idx, item_idx = self.audio_indices[
+                global_idx % len(self.audio_indices)
+            ]
+            audio_info = self.audio_lists[source_idx][item_idx]
+        else:
+            audio_info, source_idx, item_idx = util.choose_from_list_of_lists(
+                state, self.audio_lists, p=self.weights
+            )
+
+        path = audio_info["path"]
+        signal = AudioSignal.zeros(duration, sample_rate, num_channels)
+
+        if path != "none":
+            if offset is None:
+                signal = AudioSignal.salient_excerpt(
+                    path,
+                    duration=duration,
+                    state=state,
+                    loudness_cutoff=loudness_cutoff,
+                )
+            else:
+                signal = AudioSignal(
+                    path,
+                    offset=offset,
+                    duration=duration,
+                )
+
+        if num_channels == 1:
+            signal = signal.to_mono()
+        signal = signal.resample(sample_rate)
+
+        if signal.duration < duration:
+            signal = signal.zero_pad_to(int(duration * sample_rate))
+
+        for k, v in audio_info.items():
+            signal.metadata[k] = v
+
+        item = {
+            "signal": signal,
+            "source_idx": source_idx,
+            "item_idx": item_idx,
+            "source": str(self.sources[source_idx]),
+            "path": str(path),
+        }
+        if self.transform is not None:
+            item["transform_args"] = self.transform.instantiate(state, signal=signal)
+        return item
+
+
+def default_matcher(x, y):
+    return Path(x).parent == Path(y).parent
+
+
+def align_lists(lists, matcher: Callable = default_matcher):
+    longest_list = lists[np.argmax([len(l) for l in lists])]
+    for i, x in enumerate(longest_list):
+        for l in lists:
+            if i >= len(l):
+                l.append({"path": "none"})
+            elif not matcher(l[i]["path"], x["path"]):
+                l.insert(i, {"path": "none"})
+    return lists
+
+
+class AudioDataset:
+    """Loads audio from multiple loaders (with associated transforms)
+    for a specified number of samples. Excerpts are drawn randomly
+    of the specified duration, above a specified loudness threshold
+    and are resampled on the fly to the desired sample rate
+    (if it is different from the audio source sample rate).
+
+    This takes either a single AudioLoader object,
+    a dictionary of AudioLoader objects, or a dictionary of AudioLoader
+    objects. Each AudioLoader is called by the dataset, and the
+    result is placed in the output dictionary. A transform can also be
+    specified for the entire dataset, rather than for each specific
+    loader. This transform can be applied to the output of all the
+    loaders if desired.
+
+    AudioLoader objects can be specified as aligned, which means the
+    loaders correspond to multitrack audio (e.g. a vocals, bass,
+    drums, and other loader for multitrack music mixtures).
+
+
+    Parameters
+    ----------
+    loaders : Union[AudioLoader, List[AudioLoader], Dict[str, AudioLoader]]
+        AudioLoaders to sample audio from.
+    sample_rate : int
+        Desired sample rate.
+    n_examples : int, optional
+        Number of examples (length of dataset), by default 1000
+    duration : float, optional
+        Duration of audio samples, by default 0.5
+    loudness_cutoff : float, optional
+        Loudness cutoff threshold for audio samples, by default -40
+    num_channels : int, optional
+        Number of channels in output audio, by default 1
+    transform : Callable, optional
+        Transform to instantiate alongside each dataset item, by default None
+    aligned : bool, optional
+        Whether the loaders should be sampled in an aligned manner (e.g. same
+        offset, duration, and matched file name), by default False
+    shuffle_loaders : bool, optional
+        Whether to shuffle the loaders before sampling from them, by default False
+    matcher : Callable
+        How to match files from adjacent audio lists (e.g. for a multitrack audio loader),
+        by default uses the parent directory of each file.
+    without_replacement : bool
+        Whether to choose files with or without replacement, by default True.
+
+
+    Examples
+    --------
+    >>> from audiotools.data.datasets import AudioLoader
+    >>> from audiotools.data.datasets import AudioDataset
+    >>> from audiotools import transforms as tfm
+    >>> import numpy as np
+    >>>
+    >>> loaders = [
+    >>>     AudioLoader(
+    >>>         sources=[f"tests/audio/spk"],
+    >>>         transform=tfm.Equalizer(),
+    >>>         ext=["wav"],
+    >>>     )
+    >>>     for i in range(5)
+    >>> ]
+    >>>
+    >>> dataset = AudioDataset(
+    >>>     loaders = loaders,
+    >>>     sample_rate = 44100,
+    >>>     duration = 1.0,
+    >>>     transform = tfm.RescaleAudio(),
+    >>> )
+    >>>
+    >>> item = dataset[np.random.randint(len(dataset))]
+    >>>
+    >>> for i in range(len(loaders)):
+    >>>     item[i]["signal"] = loaders[i].transform(
+    >>>         item[i]["signal"], **item[i]["transform_args"]
+    >>>     )
+    >>>     item[i]["signal"].widget(i)
+    >>>
+    >>> mix = sum([item[i]["signal"] for i in range(len(loaders))])
+    >>> mix = dataset.transform(mix, **item["transform_args"])
+    >>> mix.widget("mix")
+
+    Below is an example of how one could load MUSDB multitrack data:
+
+    >>> import audiotools as at
+    >>> from pathlib import Path
+    >>> from audiotools import transforms as tfm
+    >>> import numpy as np
+    >>> import torch
+    >>>
+    >>> def build_dataset(
+    >>>     sample_rate: int = 44100,
+    >>>     duration: float = 5.0,
+    >>>     musdb_path: str = "~/.data/musdb/",
+    >>> ):
+    >>>     musdb_path = Path(musdb_path).expanduser()
+    >>>     loaders = {
+    >>>         src: at.datasets.AudioLoader(
+    >>>             sources=[musdb_path],
+    >>>             transform=tfm.Compose(
+    >>>                 tfm.VolumeNorm(("uniform", -20, -10)),
+    >>>                 tfm.Silence(prob=0.1),
+    >>>             ),
+    >>>             ext=[f"{src}.wav"],
+    >>>         )
+    >>>         for src in ["vocals", "bass", "drums", "other"]
+    >>>     }
+    >>>
+    >>>     dataset = at.datasets.AudioDataset(
+    >>>         loaders=loaders,
+    >>>         sample_rate=sample_rate,
+    >>>         duration=duration,
+    >>>         num_channels=1,
+    >>>         aligned=True,
+    >>>         transform=tfm.RescaleAudio(),
+    >>>         shuffle_loaders=True,
+    >>>     )
+    >>>     return dataset, list(loaders.keys())
+    >>>
+    >>> train_data, sources = build_dataset()
+    >>> dataloader = torch.utils.data.DataLoader(
+    >>>     train_data,
+    >>>     batch_size=16,
+    >>>     num_workers=0,
+    >>>     collate_fn=train_data.collate,
+    >>> )
+    >>> batch = next(iter(dataloader))
+    >>>
+    >>> for k in sources:
+    >>>     src = batch[k]
+    >>>     src["transformed"] = train_data.loaders[k].transform(
+    >>>         src["signal"].clone(), **src["transform_args"]
+    >>>     )
+    >>>
+    >>> mixture = sum(batch[k]["transformed"] for k in sources)
+    >>> mixture = train_data.transform(mixture, **batch["transform_args"])
+    >>>
+    >>> # Say a model takes the mix and gives back (n_batch, n_src, n_time).
+    >>> # Construct the targets:
+    >>> targets = at.AudioSignal.batch([batch[k]["transformed"] for k in sources], dim=1)
+
+    Similarly, here's example code for loading Slakh data:
+
+    >>> import audiotools as at
+    >>> from pathlib import Path
+    >>> from audiotools import transforms as tfm
+    >>> import numpy as np
+    >>> import torch
+    >>> import glob
+    >>>
+    >>> def build_dataset(
+    >>>     sample_rate: int = 16000,
+    >>>     duration: float = 10.0,
+    >>>     slakh_path: str = "~/.data/slakh/",
+    >>> ):
+    >>>     slakh_path = Path(slakh_path).expanduser()
+    >>>
+    >>>     # Find the max number of sources in Slakh
+    >>>     src_names = [x.name for x in list(slakh_path.glob("**/*.wav"))  if "S" in str(x.name)]
+    >>>     n_sources = len(list(set(src_names)))
+    >>>
+    >>>     loaders = {
+    >>>         f"S{i:02d}": at.datasets.AudioLoader(
+    >>>             sources=[slakh_path],
+    >>>             transform=tfm.Compose(
+    >>>                 tfm.VolumeNorm(("uniform", -20, -10)),
+    >>>                 tfm.Silence(prob=0.1),
+    >>>             ),
+    >>>             ext=[f"S{i:02d}.wav"],
+    >>>         )
+    >>>         for i in range(n_sources)
+    >>>     }
+    >>>     dataset = at.datasets.AudioDataset(
+    >>>         loaders=loaders,
+    >>>         sample_rate=sample_rate,
+    >>>         duration=duration,
+    >>>         num_channels=1,
+    >>>         aligned=True,
+    >>>         transform=tfm.RescaleAudio(),
+    >>>         shuffle_loaders=False,
+    >>>     )
+    >>>
+    >>>     return dataset, list(loaders.keys())
+    >>>
+    >>> train_data, sources = build_dataset()
+    >>> dataloader = torch.utils.data.DataLoader(
+    >>>     train_data,
+    >>>     batch_size=16,
+    >>>     num_workers=0,
+    >>>     collate_fn=train_data.collate,
+    >>> )
+    >>> batch = next(iter(dataloader))
+    >>>
+    >>> for k in sources:
+    >>>     src = batch[k]
+    >>>     src["transformed"] = train_data.loaders[k].transform(
+    >>>         src["signal"].clone(), **src["transform_args"]
+    >>>     )
+    >>>
+    >>> mixture = sum(batch[k]["transformed"] for k in sources)
+    >>> mixture = train_data.transform(mixture, **batch["transform_args"])
+
+    """
+
+    def __init__(
+        self,
+        loaders: Union[AudioLoader, List[AudioLoader], Dict[str, AudioLoader]],
+        sample_rate: int,
+        n_examples: int = 1000,
+        duration: float = 0.5,
+        offset: float = None,
+        loudness_cutoff: float = -40,
+        num_channels: int = 1,
+        transform: Callable = None,
+        aligned: bool = False,
+        shuffle_loaders: bool = False,
+        matcher: Callable = default_matcher,
+        without_replacement: bool = True,
+    ):
+        # Internally we convert loaders to a dictionary
+        if isinstance(loaders, list):
+            loaders = {i: l for i, l in enumerate(loaders)}
+        elif isinstance(loaders, AudioLoader):
+            loaders = {0: loaders}
+
+        self.loaders = loaders
+        self.loudness_cutoff = loudness_cutoff
+        self.num_channels = num_channels
+
+        self.length = n_examples
+        self.transform = transform
+        self.sample_rate = sample_rate
+        self.duration = duration
+        self.offset = offset
+        self.aligned = aligned
+        self.shuffle_loaders = shuffle_loaders
+        self.without_replacement = without_replacement
+
+        if aligned:
+            loaders_list = list(loaders.values())
+            for i in range(len(loaders_list[0].audio_lists)):
+                input_lists = [l.audio_lists[i] for l in loaders_list]
+                # Alignment happens in-place
+                align_lists(input_lists, matcher)
+
+    def __getitem__(self, idx):
+        state = util.random_state(idx)
+        offset = None if self.offset is None else self.offset
+        item = {}
+
+        keys = list(self.loaders.keys())
+        if self.shuffle_loaders:
+            state.shuffle(keys)
+
+        loader_kwargs = {
+            "state": state,
+            "sample_rate": self.sample_rate,
+            "duration": self.duration,
+            "loudness_cutoff": self.loudness_cutoff,
+            "num_channels": self.num_channels,
+            "global_idx": idx if self.without_replacement else None,
+        }
+
+        # Draw item from first loader
+        loader = self.loaders[keys[0]]
+        item[keys[0]] = loader(**loader_kwargs)
+
+        for key in keys[1:]:
+            loader = self.loaders[key]
+            if self.aligned:
+                # Path mapper takes the current loader + everything
+                # returned by the first loader.
+                offset = item[keys[0]]["signal"].metadata["offset"]
+                loader_kwargs.update(
+                    {
+                        "offset": offset,
+                        "source_idx": item[keys[0]]["source_idx"],
+                        "item_idx": item[keys[0]]["item_idx"],
+                    }
+                )
+            item[key] = loader(**loader_kwargs)
+
+        # Sort dictionary back into original order
+        keys = list(self.loaders.keys())
+        item = {k: item[k] for k in keys}
+
+        item["idx"] = idx
+        if self.transform is not None:
+            item["transform_args"] = self.transform.instantiate(
+                state=state, signal=item[keys[0]]["signal"]
+            )
+
+        # If there's only one loader, pop it up
+        # to the main dictionary, instead of keeping it
+        # nested.
+        if len(keys) == 1:
+            item.update(item.pop(keys[0]))
+
+        return item
+
+    def __len__(self):
+        return self.length
+
+    @staticmethod
+    def collate(list_of_dicts: Union[list, dict], n_splits: int = None):
+        """Collates items drawn from this dataset. Uses
+        :py:func:`audiotools.core.util.collate`.
+
+        Parameters
+        ----------
+        list_of_dicts : typing.Union[list, dict]
+            Data drawn from each item.
+        n_splits : int
+            Number of splits to make when creating the batches (split into
+            sub-batches). Useful for things like gradient accumulation.
+
+        Returns
+        -------
+        dict
+            Dictionary of batched data.
+        """
+        return util.collate(list_of_dicts, n_splits=n_splits)
+
+
+class ConcatDataset(AudioDataset):
+    def __init__(self, datasets: list):
+        self.datasets = datasets
+
+    def __len__(self):
+        return sum([len(d) for d in self.datasets])
+
+    def __getitem__(self, idx):
+        dataset = self.datasets[idx % len(self.datasets)]
+        return dataset[idx // len(self.datasets)]
+
+
+class ResumableDistributedSampler(DistributedSampler):  # pragma: no cover
+    """Distributed sampler that can be resumed from a given start index."""
+
+    def __init__(self, dataset, start_idx: int = None, **kwargs):
+        super().__init__(dataset, **kwargs)
+        # Start index, allows to resume an experiment at the index it was
+        self.start_idx = start_idx // self.num_replicas if start_idx is not None else 0
+
+    def __iter__(self):
+        for i, idx in enumerate(super().__iter__()):
+            if i >= self.start_idx:
+                yield idx
+        self.start_idx = 0  # set the index back to 0 so for the next epoch
+
+
+class ResumableSequentialSampler(SequentialSampler):  # pragma: no cover
+    """Sequential sampler that can be resumed from a given start index."""
+
+    def __init__(self, dataset, start_idx: int = None, **kwargs):
+        super().__init__(dataset, **kwargs)
+        # Start index, allows to resume an experiment at the index it was
+        self.start_idx = start_idx if start_idx is not None else 0
+
+    def __iter__(self):
+        for i, idx in enumerate(super().__iter__()):
+            if i >= self.start_idx:
+                yield idx
+        self.start_idx = 0  # set the index back to 0 so for the next epoch

dac-vae/audiotools/data/preprocess.py

@@ -0,0 +1,81 @@
+import csv
+import os
+from pathlib import Path
+
+from tqdm import tqdm
+
+from ..core import AudioSignal
+
+
+def create_csv(
+    audio_files: list, output_csv: Path, loudness: bool = False, data_path: str = None
+):
+    """Converts a folder of audio files to a CSV file. If ``loudness = True``,
+    the output of this function will create a CSV file that looks something
+    like:
+
+    ..  csv-table::
+        :header: path,loudness
+
+        daps/produced/f1_script1_produced.wav,-16.299999237060547
+        daps/produced/f1_script2_produced.wav,-16.600000381469727
+        daps/produced/f1_script3_produced.wav,-17.299999237060547
+        daps/produced/f1_script4_produced.wav,-16.100000381469727
+        daps/produced/f1_script5_produced.wav,-16.700000762939453
+        daps/produced/f3_script1_produced.wav,-16.5
+
+    ..  note::
+        The paths above are written relative to the ``data_path`` argument
+        which defaults to the environment variable ``PATH_TO_DATA`` if
+        it isn't passed to this function, and defaults to the empty string
+        if that environment variable is not set.
+
+    You can produce a CSV file from a directory of audio files via:
+
+    >>> import audiotools
+    >>> directory = ...
+    >>> audio_files = audiotools.util.find_audio(directory)
+    >>> output_path = "train.csv"
+    >>> audiotools.data.preprocess.create_csv(
+    >>>     audio_files, output_csv, loudness=True
+    >>> )
+
+    Note that you can create empty rows in the CSV file by passing an empty
+    string or None in the ``audio_files`` list. This is useful if you want to
+    sync multiple CSV files in a multitrack setting. The loudness of these
+    empty rows will be set to -inf.
+
+    Parameters
+    ----------
+    audio_files : list
+        List of audio files.
+    output_csv : Path
+        Output CSV, with each row containing the relative path of every file
+        to ``data_path``, if specified (defaults to None).
+    loudness : bool
+        Compute loudness of entire file and store alongside path.
+    """
+
+    info = []
+    pbar = tqdm(audio_files)
+    for af in pbar:
+        af = Path(af)
+        pbar.set_description(f"Processing {af.name}")
+        _info = {}
+        if af.name == "":
+            _info["path"] = ""
+            if loudness:
+                _info["loudness"] = -float("inf")
+        else:
+            _info["path"] = af.relative_to(data_path) if data_path is not None else af
+            if loudness:
+                _info["loudness"] = AudioSignal(af).ffmpeg_loudness().item()
+
+        info.append(_info)
+
+    with open(output_csv, "w") as f:
+        writer = csv.DictWriter(f, fieldnames=list(info[0].keys()))
+        writer.writeheader()
+
+        for item in info:
+            writer.writerow(item)

dac-vae/audiotools/data/transforms.py

@@ -0,0 +1,1592 @@
+import copy
+from contextlib import contextmanager
+from inspect import signature
+from typing import List
+
+import numpy as np
+import torch
+from flatten_dict import flatten
+from flatten_dict import unflatten
+from numpy.random import RandomState
+
+from .. import ml
+from ..core import AudioSignal
+from ..core import util
+from .datasets import AudioLoader
+
+tt = torch.tensor
+"""Shorthand for converting things to torch.tensor."""
+
+
+class BaseTransform:
+    """This is the base class for all transforms that are implemented
+    in this library. Transforms have two main operations: ``transform``
+    and ``instantiate``.
+
+    ``instantiate`` sets the parameters randomly
+    from distribution tuples for each parameter. For example, for the
+    ``BackgroundNoise`` transform, the signal-to-noise ratio (``snr``)
+    is chosen randomly by instantiate. By default, it chosen uniformly
+    between 10.0 and 30.0 (the tuple is set to ``("uniform", 10.0, 30.0)``).
+
+    ``transform`` applies the transform using the instantiated parameters.
+    A simple example is as follows:
+
+    >>> seed = 0
+    >>> signal = ...
+    >>> transform = transforms.NoiseFloor(db = ("uniform", -50.0, -30.0))
+    >>> kwargs = transform.instantiate()
+    >>> output = transform(signal.clone(), **kwargs)
+
+    By breaking apart the instantiation of parameters from the actual audio
+    processing of the transform, we can make things more reproducible, while
+    also applying the transform on batches of data efficiently on GPU,
+    rather than on individual audio samples.
+
+    ..  note::
+        We call ``signal.clone()`` for the input to the ``transform`` function
+        because signals are modified in-place! If you don't clone the signal,
+        you will lose the original data.
+
+    Parameters
+    ----------
+    keys : list, optional
+        Keys that the transform looks for when
+        calling ``self.transform``, by default []. In general this is
+        set automatically, and you won't need to manipulate this argument.
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+
+    Examples
+    --------
+
+    >>> seed = 0
+    >>>
+    >>> audio_path = "tests/audio/spk/f10_script4_produced.wav"
+    >>> signal = AudioSignal(audio_path, offset=10, duration=2)
+    >>> transform = tfm.Compose(
+    >>>     [
+    >>>         tfm.RoomImpulseResponse(sources=["tests/audio/irs.csv"]),
+    >>>         tfm.BackgroundNoise(sources=["tests/audio/noises.csv"]),
+    >>>     ],
+    >>> )
+    >>>
+    >>> kwargs = transform.instantiate(seed, signal)
+    >>> output = transform(signal, **kwargs)
+
+    """
+
+    def __init__(self, keys: list = [], name: str = None, prob: float = 1.0):
+        # Get keys from the _transform signature.
+        tfm_keys = list(signature(self._transform).parameters.keys())
+
+        # Filter out signal and kwargs keys.
+        ignore_keys = ["signal", "kwargs"]
+        tfm_keys = [k for k in tfm_keys if k not in ignore_keys]
+
+        # Combine keys specified by the child class, the keys found in
+        # _transform signature, and the mask key.
+        self.keys = keys + tfm_keys + ["mask"]
+
+        self.prob = prob
+
+        if name is None:
+            name = self.__class__.__name__
+        self.name = name
+
+    def _prepare(self, batch: dict):
+        sub_batch = batch[self.name]
+
+        for k in self.keys:
+            assert k in sub_batch.keys(), f"{k} not in batch"
+
+        return sub_batch
+
+    def _transform(self, signal):
+        return signal
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        return {}
+
+    @staticmethod
+    def apply_mask(batch: dict, mask: torch.Tensor):
+        """Applies a mask to the batch.
+
+        Parameters
+        ----------
+        batch : dict
+            Batch whose values will be masked in the ``transform`` pass.
+        mask : torch.Tensor
+            Mask to apply to batch.
+
+        Returns
+        -------
+        dict
+            A dictionary that contains values only where ``mask = True``.
+        """
+        masked_batch = {k: v[mask] for k, v in flatten(batch).items()}
+        return unflatten(masked_batch)
+
+    def transform(self, signal: AudioSignal, **kwargs):
+        """Apply the transform to the audio signal,
+        with given keyword arguments.
+
+        Parameters
+        ----------
+        signal : AudioSignal
+            Signal that will be modified by the transforms in-place.
+        kwargs: dict
+            Keyword arguments to the specific transforms ``self._transform``
+            function.
+
+        Returns
+        -------
+        AudioSignal
+            Transformed AudioSignal.
+
+        Examples
+        --------
+
+        >>> for seed in range(10):
+        >>>     kwargs = transform.instantiate(seed, signal)
+        >>>     output = transform(signal.clone(), **kwargs)
+
+        """
+        tfm_kwargs = self._prepare(kwargs)
+        mask = tfm_kwargs["mask"]
+
+        if torch.any(mask):
+            tfm_kwargs = self.apply_mask(tfm_kwargs, mask)
+            tfm_kwargs = {k: v for k, v in tfm_kwargs.items() if k != "mask"}
+            signal[mask] = self._transform(signal[mask], **tfm_kwargs)
+
+        return signal
+
+    def __call__(self, *args, **kwargs):
+        return self.transform(*args, **kwargs)
+
+    def instantiate(
+        self,
+        state: RandomState = None,
+        signal: AudioSignal = None,
+    ):
+        """Instantiates parameters for the transform.
+
+        Parameters
+        ----------
+        state : RandomState, optional
+            _description_, by default None
+        signal : AudioSignal, optional
+            _description_, by default None
+
+        Returns
+        -------
+        dict
+            Dictionary containing instantiated arguments for every keyword
+            argument to ``self._transform``.
+
+        Examples
+        --------
+
+        >>> for seed in range(10):
+        >>>     kwargs = transform.instantiate(seed, signal)
+        >>>     output = transform(signal.clone(), **kwargs)
+
+        """
+        state = util.random_state(state)
+
+        # Not all instantiates need the signal. Check if signal
+        # is needed before passing it in, so that the end-user
+        # doesn't need to have variables they're not using flowing
+        # into their function.
+        needs_signal = "signal" in set(signature(self._instantiate).parameters.keys())
+        kwargs = {}
+        if needs_signal:
+            kwargs = {"signal": signal}
+
+        # Instantiate the parameters for the transform.
+        params = self._instantiate(state, **kwargs)
+        for k in list(params.keys()):
+            v = params[k]
+            if isinstance(v, (AudioSignal, torch.Tensor, dict)):
+                params[k] = v
+            else:
+                params[k] = tt(v)
+        mask = state.rand() <= self.prob
+        params[f"mask"] = tt(mask)
+
+        # Put the params into a nested dictionary that will be
+        # used later when calling the transform. This is to avoid
+        # collisions in the dictionary.
+        params = {self.name: params}
+
+        return params
+
+    def batch_instantiate(
+        self,
+        states: list = None,
+        signal: AudioSignal = None,
+    ):
+        """Instantiates arguments for every item in a batch,
+        given a list of states. Each state in the list
+        corresponds to one item in the batch.
+
+        Parameters
+        ----------
+        states : list, optional
+            List of states, by default None
+        signal : AudioSignal, optional
+            AudioSignal to pass to the ``self.instantiate`` section
+            if it is needed for this transform, by default None
+
+        Returns
+        -------
+        dict
+            Collated dictionary of arguments.
+
+        Examples
+        --------
+
+        >>> batch_size = 4
+        >>> signal = AudioSignal(audio_path, offset=10, duration=2)
+        >>> signal_batch = AudioSignal.batch([signal.clone() for _ in range(batch_size)])
+        >>>
+        >>> states = [seed + idx for idx in list(range(batch_size))]
+        >>> kwargs = transform.batch_instantiate(states, signal_batch)
+        >>> batch_output = transform(signal_batch, **kwargs)
+        """
+        kwargs = []
+        for state in states:
+            kwargs.append(self.instantiate(state, signal))
+        kwargs = util.collate(kwargs)
+        return kwargs
+
+
+class Identity(BaseTransform):
+    """This transform just returns the original signal."""
+
+    pass
+
+
+class SpectralTransform(BaseTransform):
+    """Spectral transforms require STFT data to exist, since manipulations
+    of the STFT require the spectrogram. This just calls ``stft`` before
+    the transform is called, and calls ``istft`` after the transform is
+    called so that the audio data is written to after the spectral
+    manipulation.
+    """
+
+    def transform(self, signal, **kwargs):
+        signal.stft()
+        super().transform(signal, **kwargs)
+        signal.istft()
+        return signal
+
+
+class Compose(BaseTransform):
+    """Compose applies transforms in sequence, one after the other. The
+    transforms are passed in as positional arguments or as a list like so:
+
+    >>> transform = tfm.Compose(
+    >>>     [
+    >>>         tfm.RoomImpulseResponse(sources=["tests/audio/irs.csv"]),
+    >>>         tfm.BackgroundNoise(sources=["tests/audio/noises.csv"]),
+    >>>     ],
+    >>> )
+
+    This will convolve the signal with a room impulse response, and then
+    add background noise to the signal. Instantiate instantiates
+    all the parameters for every transform in the transform list so the
+    interface for using the Compose transform is the same as everything
+    else:
+
+    >>> kwargs = transform.instantiate()
+    >>> output = transform(signal.clone(), **kwargs)
+
+    Under the hood, the transform maps each transform to a unique name
+    under the hood of the form ``{position}.{name}``, where ``position``
+    is the index of the transform in the list. ``Compose`` can nest
+    within other ``Compose`` transforms, like so:
+
+    >>> preprocess = transforms.Compose(
+    >>>     tfm.GlobalVolumeNorm(),
+    >>>     tfm.CrossTalk(),
+    >>>     name="preprocess",
+    >>> )
+    >>> augment = transforms.Compose(
+    >>>     tfm.RoomImpulseResponse(),
+    >>>     tfm.BackgroundNoise(),
+    >>>     name="augment",
+    >>> )
+    >>> postprocess = transforms.Compose(
+    >>>     tfm.VolumeChange(),
+    >>>     tfm.RescaleAudio(),
+    >>>     tfm.ShiftPhase(),
+    >>>     name="postprocess",
+    >>> )
+    >>> transform = transforms.Compose(preprocess, augment, postprocess),
+
+    This defines 3 composed transforms, and then composes them in sequence
+    with one another.
+
+    Parameters
+    ----------
+    *transforms : list
+        List of transforms to apply
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(self, *transforms: list, name: str = None, prob: float = 1.0):
+        if isinstance(transforms[0], list):
+            transforms = transforms[0]
+
+        for i, tfm in enumerate(transforms):
+            tfm.name = f"{i}.{tfm.name}"
+
+        keys = [tfm.name for tfm in transforms]
+        super().__init__(keys=keys, name=name, prob=prob)
+
+        self.transforms = transforms
+        self.transforms_to_apply = keys
+
+    @contextmanager
+    def filter(self, *names: list):
+        """This can be used to skip transforms entirely when applying
+        the sequence of transforms to a signal. For example, take
+        the following transforms with the names ``preprocess, augment, postprocess``.
+
+        >>> preprocess = transforms.Compose(
+        >>>     tfm.GlobalVolumeNorm(),
+        >>>     tfm.CrossTalk(),
+        >>>     name="preprocess",
+        >>> )
+        >>> augment = transforms.Compose(
+        >>>     tfm.RoomImpulseResponse(),
+        >>>     tfm.BackgroundNoise(),
+        >>>     name="augment",
+        >>> )
+        >>> postprocess = transforms.Compose(
+        >>>     tfm.VolumeChange(),
+        >>>     tfm.RescaleAudio(),
+        >>>     tfm.ShiftPhase(),
+        >>>     name="postprocess",
+        >>> )
+        >>> transform = transforms.Compose(preprocess, augment, postprocess)
+
+        If we wanted to apply all 3 to a signal, we do:
+
+        >>> kwargs = transform.instantiate()
+        >>> output = transform(signal.clone(), **kwargs)
+
+        But if we only wanted to apply the ``preprocess`` and ``postprocess``
+        transforms to the signal, we do:
+
+        >>> with transform_fn.filter("preprocess", "postprocess"):
+        >>>     output = transform(signal.clone(), **kwargs)
+
+        Parameters
+        ----------
+        *names : list
+            List of transforms, identified by name, to apply to signal.
+        """
+        old_transforms = self.transforms_to_apply
+        self.transforms_to_apply = names
+        yield
+        self.transforms_to_apply = old_transforms
+
+    def _transform(self, signal, **kwargs):
+        for transform in self.transforms:
+            if any([x in transform.name for x in self.transforms_to_apply]):
+                signal = transform(signal, **kwargs)
+        return signal
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        parameters = {}
+        for transform in self.transforms:
+            parameters.update(transform.instantiate(state, signal=signal))
+        return parameters
+
+    def __getitem__(self, idx):
+        return self.transforms[idx]
+
+    def __len__(self):
+        return len(self.transforms)
+
+    def __iter__(self):
+        for transform in self.transforms:
+            yield transform
+
+
+class Choose(Compose):
+    """Choose logic is the same as :py:func:`audiotools.data.transforms.Compose`,
+    but instead of applying all the transforms in sequence, it applies just a single transform,
+    which is chosen for each item in the batch.
+
+    Parameters
+    ----------
+    *transforms : list
+        List of transforms to apply
+    weights : list
+        Probability of choosing any specific transform.
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+
+    Examples
+    --------
+
+    >>> transforms.Choose(tfm.LowPass(), tfm.HighPass())
+    """
+
+    def __init__(
+        self,
+        *transforms: list,
+        weights: list = None,
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(*transforms, name=name, prob=prob)
+
+        if weights is None:
+            _len = len(self.transforms)
+            weights = [1 / _len for _ in range(_len)]
+        self.weights = np.array(weights)
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        kwargs = super()._instantiate(state, signal)
+        tfm_idx = list(range(len(self.transforms)))
+        tfm_idx = state.choice(tfm_idx, p=self.weights)
+        one_hot = []
+        for i, t in enumerate(self.transforms):
+            mask = kwargs[t.name]["mask"]
+            if mask.item():
+                kwargs[t.name]["mask"] = tt(i == tfm_idx)
+            one_hot.append(kwargs[t.name]["mask"])
+        kwargs["one_hot"] = one_hot
+        return kwargs
+
+
+class Repeat(Compose):
+    """Repeatedly applies a given transform ``n_repeat`` times."
+
+    Parameters
+    ----------
+    transform : BaseTransform
+        Transform to repeat.
+    n_repeat : int, optional
+        Number of times to repeat transform, by default 1
+    """
+
+    def __init__(
+        self,
+        transform,
+        n_repeat: int = 1,
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        transforms = [copy.copy(transform) for _ in range(n_repeat)]
+        super().__init__(transforms, name=name, prob=prob)
+
+        self.n_repeat = n_repeat
+
+
+class RepeatUpTo(Choose):
+    """Repeatedly applies a given transform up to ``max_repeat`` times."
+
+    Parameters
+    ----------
+    transform : BaseTransform
+        Transform to repeat.
+    max_repeat : int, optional
+        Max number of times to repeat transform, by default 1
+    weights : list
+        Probability of choosing any specific number up to ``max_repeat``.
+    """
+
+    def __init__(
+        self,
+        transform,
+        max_repeat: int = 5,
+        weights: list = None,
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        transforms = []
+        for n in range(1, max_repeat):
+            transforms.append(Repeat(transform, n_repeat=n))
+        super().__init__(transforms, name=name, prob=prob, weights=weights)
+
+        self.max_repeat = max_repeat
+
+
+class ClippingDistortion(BaseTransform):
+    """Adds clipping distortion to signal. Corresponds
+    to :py:func:`audiotools.core.effects.EffectMixin.clip_distortion`.
+
+    Parameters
+    ----------
+    perc : tuple, optional
+        Clipping percentile. Values are between 0.0 to 1.0.
+        Typical values are 0.1 or below, by default ("uniform", 0.0, 0.1)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        perc: tuple = ("uniform", 0.0, 0.1),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.perc = perc
+
+    def _instantiate(self, state: RandomState):
+        return {"perc": util.sample_from_dist(self.perc, state)}
+
+    def _transform(self, signal, perc):
+        return signal.clip_distortion(perc)
+
+
+class Equalizer(BaseTransform):
+    """Applies an equalization curve to the audio signal. Corresponds
+    to :py:func:`audiotools.core.effects.EffectMixin.equalizer`.
+
+    Parameters
+    ----------
+    eq_amount : tuple, optional
+        The maximum dB cut to apply to the audio in any band,
+        by default ("const", 1.0 dB)
+    n_bands : int, optional
+        Number of bands in EQ, by default 6
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        eq_amount: tuple = ("const", 1.0),
+        n_bands: int = 6,
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.eq_amount = eq_amount
+        self.n_bands = n_bands
+
+    def _instantiate(self, state: RandomState):
+        eq_amount = util.sample_from_dist(self.eq_amount, state)
+        eq = -eq_amount * state.rand(self.n_bands)
+        return {"eq": eq}
+
+    def _transform(self, signal, eq):
+        return signal.equalizer(eq)
+
+
+class Quantization(BaseTransform):
+    """Applies quantization to the input waveform. Corresponds
+    to :py:func:`audiotools.core.effects.EffectMixin.quantization`.
+
+    Parameters
+    ----------
+    channels : tuple, optional
+        Number of evenly spaced quantization channels to quantize
+        to, by default ("choice", [8, 32, 128, 256, 1024])
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        channels: tuple = ("choice", [8, 32, 128, 256, 1024]),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.channels = channels
+
+    def _instantiate(self, state: RandomState):
+        return {"channels": util.sample_from_dist(self.channels, state)}
+
+    def _transform(self, signal, channels):
+        return signal.quantization(channels)
+
+
+class MuLawQuantization(BaseTransform):
+    """Applies mu-law quantization to the input waveform. Corresponds
+    to :py:func:`audiotools.core.effects.EffectMixin.mulaw_quantization`.
+
+    Parameters
+    ----------
+    channels : tuple, optional
+        Number of mu-law spaced quantization channels to quantize
+        to, by default ("choice", [8, 32, 128, 256, 1024])
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        channels: tuple = ("choice", [8, 32, 128, 256, 1024]),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.channels = channels
+
+    def _instantiate(self, state: RandomState):
+        return {"channels": util.sample_from_dist(self.channels, state)}
+
+    def _transform(self, signal, channels):
+        return signal.mulaw_quantization(channels)
+
+
+class NoiseFloor(BaseTransform):
+    """Adds a noise floor of Gaussian noise to the signal at a specified
+    dB.
+
+    Parameters
+    ----------
+    db : tuple, optional
+        Level of noise to add to signal, by default ("const", -50.0)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        db: tuple = ("const", -50.0),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.db = db
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        db = util.sample_from_dist(self.db, state)
+        audio_data = state.randn(signal.num_channels, signal.signal_length)
+        nz_signal = AudioSignal(audio_data, signal.sample_rate)
+        nz_signal.normalize(db)
+        return {"nz_signal": nz_signal}
+
+    def _transform(self, signal, nz_signal):
+        # Clone bg_signal so that transform can be repeatedly applied
+        # to different signals with the same effect.
+        return signal + nz_signal
+
+
+class BackgroundNoise(BaseTransform):
+    """Adds background noise from audio specified by a set of CSV files.
+    A valid CSV file looks like, and is typically generated by
+    :py:func:`audiotools.data.preprocess.create_csv`:
+
+    ..  csv-table::
+        :header: path
+
+        room_tone/m6_script2_clean.wav
+        room_tone/m6_script2_cleanraw.wav
+        room_tone/m6_script2_ipad_balcony1.wav
+        room_tone/m6_script2_ipad_bedroom1.wav
+        room_tone/m6_script2_ipad_confroom1.wav
+        room_tone/m6_script2_ipad_confroom2.wav
+        room_tone/m6_script2_ipad_livingroom1.wav
+        room_tone/m6_script2_ipad_office1.wav
+
+    ..  note::
+        All paths are relative to an environment variable called ``PATH_TO_DATA``,
+        so that CSV files are portable across machines where data may be
+        located in different places.
+
+    This transform calls :py:func:`audiotools.core.effects.EffectMixin.mix`
+    and :py:func:`audiotools.core.effects.EffectMixin.equalizer` under the
+    hood.
+
+    Parameters
+    ----------
+    snr : tuple, optional
+        Signal-to-noise ratio, by default ("uniform", 10.0, 30.0)
+    sources : List[str], optional
+        Sources containing folders, or CSVs with paths to audio files,
+        by default None
+    weights : List[float], optional
+        Weights to sample audio files from each source, by default None
+    eq_amount : tuple, optional
+        Amount of equalization to apply, by default ("const", 1.0)
+    n_bands : int, optional
+        Number of bands in equalizer, by default 3
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    loudness_cutoff : float, optional
+        Loudness cutoff when loading from audio files, by default None
+    """
+
+    def __init__(
+        self,
+        snr: tuple = ("uniform", 10.0, 30.0),
+        sources: List[str] = None,
+        weights: List[float] = None,
+        eq_amount: tuple = ("const", 1.0),
+        n_bands: int = 3,
+        name: str = None,
+        prob: float = 1.0,
+        loudness_cutoff: float = None,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.snr = snr
+        self.eq_amount = eq_amount
+        self.n_bands = n_bands
+        self.loader = AudioLoader(sources, weights)
+        self.loudness_cutoff = loudness_cutoff
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        eq_amount = util.sample_from_dist(self.eq_amount, state)
+        eq = -eq_amount * state.rand(self.n_bands)
+        snr = util.sample_from_dist(self.snr, state)
+
+        bg_signal = self.loader(
+            state,
+            signal.sample_rate,
+            duration=signal.signal_duration,
+            loudness_cutoff=self.loudness_cutoff,
+            num_channels=signal.num_channels,
+        )["signal"]
+
+        return {"eq": eq, "bg_signal": bg_signal, "snr": snr}
+
+    def _transform(self, signal, bg_signal, snr, eq):
+        # Clone bg_signal so that transform can be repeatedly applied
+        # to different signals with the same effect.
+        return signal.mix(bg_signal.clone(), snr, eq)
+
+
+class CrossTalk(BaseTransform):
+    """Adds crosstalk between speakers, whose audio is drawn from a CSV file
+    that was produced via :py:func:`audiotools.data.preprocess.create_csv`.
+
+    This transform calls :py:func:`audiotools.core.effects.EffectMixin.mix`
+    under the hood.
+
+    Parameters
+    ----------
+    snr : tuple, optional
+        How loud cross-talk speaker is relative to original signal in dB,
+        by default ("uniform", 0.0, 10.0)
+    sources : List[str], optional
+        Sources containing folders, or CSVs with paths to audio files,
+        by default None
+    weights : List[float], optional
+        Weights to sample audio files from each source, by default None
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    loudness_cutoff : float, optional
+        Loudness cutoff when loading from audio files, by default -40
+    """
+
+    def __init__(
+        self,
+        snr: tuple = ("uniform", 0.0, 10.0),
+        sources: List[str] = None,
+        weights: List[float] = None,
+        name: str = None,
+        prob: float = 1.0,
+        loudness_cutoff: float = -40,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.snr = snr
+        self.loader = AudioLoader(sources, weights)
+        self.loudness_cutoff = loudness_cutoff
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        snr = util.sample_from_dist(self.snr, state)
+        crosstalk_signal = self.loader(
+            state,
+            signal.sample_rate,
+            duration=signal.signal_duration,
+            loudness_cutoff=self.loudness_cutoff,
+            num_channels=signal.num_channels,
+        )["signal"]
+
+        return {"crosstalk_signal": crosstalk_signal, "snr": snr}
+
+    def _transform(self, signal, crosstalk_signal, snr):
+        # Clone bg_signal so that transform can be repeatedly applied
+        # to different signals with the same effect.
+        loudness = signal.loudness()
+        mix = signal.mix(crosstalk_signal.clone(), snr)
+        mix.normalize(loudness)
+        return mix
+
+
+class RoomImpulseResponse(BaseTransform):
+    """Convolves signal with a room impulse response, at a specified
+    direct-to-reverberant ratio, with equalization applied. Room impulse
+    response data is drawn from a CSV file that was produced via
+    :py:func:`audiotools.data.preprocess.create_csv`.
+
+    This transform calls :py:func:`audiotools.core.effects.EffectMixin.apply_ir`
+    under the hood.
+
+    Parameters
+    ----------
+    drr : tuple, optional
+        _description_, by default ("uniform", 0.0, 30.0)
+    sources : List[str], optional
+        Sources containing folders, or CSVs with paths to audio files,
+        by default None
+    weights : List[float], optional
+        Weights to sample audio files from each source, by default None
+    eq_amount : tuple, optional
+        Amount of equalization to apply, by default ("const", 1.0)
+    n_bands : int, optional
+        Number of bands in equalizer, by default 6
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    use_original_phase : bool, optional
+        Whether or not to use the original phase, by default False
+    offset : float, optional
+        Offset from each impulse response file to use, by default 0.0
+    duration : float, optional
+        Duration of each impulse response, by default 1.0
+    """
+
+    def __init__(
+        self,
+        drr: tuple = ("uniform", 0.0, 30.0),
+        sources: List[str] = None,
+        weights: List[float] = None,
+        eq_amount: tuple = ("const", 1.0),
+        n_bands: int = 6,
+        name: str = None,
+        prob: float = 1.0,
+        use_original_phase: bool = False,
+        offset: float = 0.0,
+        duration: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.drr = drr
+        self.eq_amount = eq_amount
+        self.n_bands = n_bands
+        self.use_original_phase = use_original_phase
+
+        self.loader = AudioLoader(sources, weights)
+        self.offset = offset
+        self.duration = duration
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        eq_amount = util.sample_from_dist(self.eq_amount, state)
+        eq = -eq_amount * state.rand(self.n_bands)
+        drr = util.sample_from_dist(self.drr, state)
+
+        ir_signal = self.loader(
+            state,
+            signal.sample_rate,
+            offset=self.offset,
+            duration=self.duration,
+            loudness_cutoff=None,
+            num_channels=signal.num_channels,
+        )["signal"]
+        ir_signal.zero_pad_to(signal.sample_rate)
+
+        return {"eq": eq, "ir_signal": ir_signal, "drr": drr}
+
+    def _transform(self, signal, ir_signal, drr, eq):
+        # Clone ir_signal so that transform can be repeatedly applied
+        # to different signals with the same effect.
+        return signal.apply_ir(
+            ir_signal.clone(), drr, eq, use_original_phase=self.use_original_phase
+        )
+
+
+class VolumeChange(BaseTransform):
+    """Changes the volume of the input signal.
+
+    Uses :py:func:`audiotools.core.effects.EffectMixin.volume_change`.
+
+    Parameters
+    ----------
+    db : tuple, optional
+        Change in volume in decibels, by default ("uniform", -12.0, 0.0)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        db: tuple = ("uniform", -12.0, 0.0),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.db = db
+
+    def _instantiate(self, state: RandomState):
+        return {"db": util.sample_from_dist(self.db, state)}
+
+    def _transform(self, signal, db):
+        return signal.volume_change(db)
+
+
+class VolumeNorm(BaseTransform):
+    """Normalizes the volume of the excerpt to a specified decibel.
+
+    Uses :py:func:`audiotools.core.effects.EffectMixin.normalize`.
+
+    Parameters
+    ----------
+    db : tuple, optional
+        dB to normalize signal to, by default ("const", -24)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        db: tuple = ("const", -24),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.db = db
+
+    def _instantiate(self, state: RandomState):
+        return {"db": util.sample_from_dist(self.db, state)}
+
+    def _transform(self, signal, db):
+        return signal.normalize(db)
+
+
+class GlobalVolumeNorm(BaseTransform):
+    """Similar to :py:func:`audiotools.data.transforms.VolumeNorm`, this
+    transform also normalizes the volume of a signal, but it uses
+    the volume of the entire audio file the loaded excerpt comes from,
+    rather than the volume of just the excerpt. The volume of the
+    entire audio file is expected in ``signal.metadata["loudness"]``.
+    If loading audio from a CSV generated by :py:func:`audiotools.data.preprocess.create_csv`
+    with ``loudness = True``, like the following:
+
+    ..  csv-table::
+        :header: path,loudness
+
+        daps/produced/f1_script1_produced.wav,-16.299999237060547
+        daps/produced/f1_script2_produced.wav,-16.600000381469727
+        daps/produced/f1_script3_produced.wav,-17.299999237060547
+        daps/produced/f1_script4_produced.wav,-16.100000381469727
+        daps/produced/f1_script5_produced.wav,-16.700000762939453
+        daps/produced/f3_script1_produced.wav,-16.5
+
+    The ``AudioLoader`` will automatically load the loudness column into
+    the metadata of the signal.
+
+    Uses :py:func:`audiotools.core.effects.EffectMixin.volume_change`.
+
+    Parameters
+    ----------
+    db : tuple, optional
+        dB to normalize signal to, by default ("const", -24)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        db: tuple = ("const", -24),
+        name: str = None,
+        prob: float = 1.0,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.db = db
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        if "loudness" not in signal.metadata:
+            db_change = 0.0
+        elif float(signal.metadata["loudness"]) == float("-inf"):
+            db_change = 0.0
+        else:
+            db = util.sample_from_dist(self.db, state)
+            db_change = db - float(signal.metadata["loudness"])
+
+        return {"db": db_change}
+
+    def _transform(self, signal, db):
+        return signal.volume_change(db)
+
+
+class Silence(BaseTransform):
+    """Zeros out the signal with some probability.
+
+    Parameters
+    ----------
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 0.1
+    """
+
+    def __init__(self, name: str = None, prob: float = 0.1):
+        super().__init__(name=name, prob=prob)
+
+    def _transform(self, signal):
+        _loudness = signal._loudness
+        signal = AudioSignal(
+            torch.zeros_like(signal.audio_data),
+            sample_rate=signal.sample_rate,
+            stft_params=signal.stft_params,
+        )
+        # So that the amound of noise added is as if it wasn't silenced.
+        # TODO: improve this hack
+        signal._loudness = _loudness
+
+        return signal
+
+
+class LowPass(BaseTransform):
+    """Applies a LowPass filter.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.low_pass`.
+
+    Parameters
+    ----------
+    cutoff : tuple, optional
+        Cutoff frequency distribution,
+        by default ``("choice", [4000, 8000, 16000])``
+    zeros : int, optional
+        Number of zero-crossings in filter, argument to
+        ``julius.LowPassFilters``, by default 51
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        cutoff: tuple = ("choice", [4000, 8000, 16000]),
+        zeros: int = 51,
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.cutoff = cutoff
+        self.zeros = zeros
+
+    def _instantiate(self, state: RandomState):
+        return {"cutoff": util.sample_from_dist(self.cutoff, state)}
+
+    def _transform(self, signal, cutoff):
+        return signal.low_pass(cutoff, zeros=self.zeros)
+
+
+class HighPass(BaseTransform):
+    """Applies a HighPass filter.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.high_pass`.
+
+    Parameters
+    ----------
+    cutoff : tuple, optional
+        Cutoff frequency distribution,
+        by default ``("choice", [50, 100, 250, 500, 1000])``
+    zeros : int, optional
+        Number of zero-crossings in filter, argument to
+        ``julius.LowPassFilters``, by default 51
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        cutoff: tuple = ("choice", [50, 100, 250, 500, 1000]),
+        zeros: int = 51,
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+
+        self.cutoff = cutoff
+        self.zeros = zeros
+
+    def _instantiate(self, state: RandomState):
+        return {"cutoff": util.sample_from_dist(self.cutoff, state)}
+
+    def _transform(self, signal, cutoff):
+        return signal.high_pass(cutoff, zeros=self.zeros)
+
+
+class RescaleAudio(BaseTransform):
+    """Rescales the audio so it is in between ``-val`` and ``val``
+    only if the original audio exceeds those bounds. Useful if
+    transforms have caused the audio to clip.
+
+    Uses :py:func:`audiotools.core.effects.EffectMixin.ensure_max_of_audio`.
+
+    Parameters
+    ----------
+    val : float, optional
+        Max absolute value of signal, by default 1.0
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(self, val: float = 1.0, name: str = None, prob: float = 1):
+        super().__init__(name=name, prob=prob)
+
+        self.val = val
+
+    def _transform(self, signal):
+        return signal.ensure_max_of_audio(self.val)
+
+
+class ShiftPhase(SpectralTransform):
+    """Shifts the phase of the audio.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.shift)phase`.
+
+    Parameters
+    ----------
+    shift : tuple, optional
+        How much to shift phase by, by default ("uniform", -np.pi, np.pi)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        shift: tuple = ("uniform", -np.pi, np.pi),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.shift = shift
+
+    def _instantiate(self, state: RandomState):
+        return {"shift": util.sample_from_dist(self.shift, state)}
+
+    def _transform(self, signal, shift):
+        return signal.shift_phase(shift)
+
+
+class InvertPhase(ShiftPhase):
+    """Inverts the phase of the audio.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.shift_phase`.
+
+    Parameters
+    ----------
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(self, name: str = None, prob: float = 1):
+        super().__init__(shift=("const", np.pi), name=name, prob=prob)
+
+
+class CorruptPhase(SpectralTransform):
+    """Corrupts the phase of the audio.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.corrupt_phase`.
+
+    Parameters
+    ----------
+    scale : tuple, optional
+        How much to corrupt phase by, by default ("uniform", 0, np.pi)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self, scale: tuple = ("uniform", 0, np.pi), name: str = None, prob: float = 1
+    ):
+        super().__init__(name=name, prob=prob)
+        self.scale = scale
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        scale = util.sample_from_dist(self.scale, state)
+        corruption = state.normal(scale=scale, size=signal.phase.shape[1:])
+        return {"corruption": corruption.astype("float32")}
+
+    def _transform(self, signal, corruption):
+        return signal.shift_phase(shift=corruption)
+
+
+class FrequencyMask(SpectralTransform):
+    """Masks a band of frequencies at a center frequency
+    from the audio.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.mask_frequencies`.
+
+    Parameters
+    ----------
+    f_center : tuple, optional
+        Center frequency between 0.0 and 1.0 (Nyquist), by default ("uniform", 0.0, 1.0)
+    f_width : tuple, optional
+        Width of zero'd out band, by default ("const", 0.1)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        f_center: tuple = ("uniform", 0.0, 1.0),
+        f_width: tuple = ("const", 0.1),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.f_center = f_center
+        self.f_width = f_width
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        f_center = util.sample_from_dist(self.f_center, state)
+        f_width = util.sample_from_dist(self.f_width, state)
+
+        fmin = max(f_center - (f_width / 2), 0.0)
+        fmax = min(f_center + (f_width / 2), 1.0)
+
+        fmin_hz = (signal.sample_rate / 2) * fmin
+        fmax_hz = (signal.sample_rate / 2) * fmax
+
+        return {"fmin_hz": fmin_hz, "fmax_hz": fmax_hz}
+
+    def _transform(self, signal, fmin_hz: float, fmax_hz: float):
+        return signal.mask_frequencies(fmin_hz=fmin_hz, fmax_hz=fmax_hz)
+
+
+class TimeMask(SpectralTransform):
+    """Masks out contiguous time-steps from signal.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.mask_timesteps`.
+
+    Parameters
+    ----------
+    t_center : tuple, optional
+        Center time in terms of 0.0 and 1.0 (duration of signal),
+        by default ("uniform", 0.0, 1.0)
+    t_width : tuple, optional
+        Width of dropped out portion, by default ("const", 0.025)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        t_center: tuple = ("uniform", 0.0, 1.0),
+        t_width: tuple = ("const", 0.025),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.t_center = t_center
+        self.t_width = t_width
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal):
+        t_center = util.sample_from_dist(self.t_center, state)
+        t_width = util.sample_from_dist(self.t_width, state)
+
+        tmin = max(t_center - (t_width / 2), 0.0)
+        tmax = min(t_center + (t_width / 2), 1.0)
+
+        tmin_s = signal.signal_duration * tmin
+        tmax_s = signal.signal_duration * tmax
+        return {"tmin_s": tmin_s, "tmax_s": tmax_s}
+
+    def _transform(self, signal, tmin_s: float, tmax_s: float):
+        return signal.mask_timesteps(tmin_s=tmin_s, tmax_s=tmax_s)
+
+
+class MaskLowMagnitudes(SpectralTransform):
+    """Masks low magnitude regions out of signal.
+
+    Uses :py:func:`audiotools.core.dsp.DSPMixin.mask_low_magnitudes`.
+
+    Parameters
+    ----------
+    db_cutoff : tuple, optional
+        Decibel value for which things below it will be masked away,
+        by default ("uniform", -10, 10)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        db_cutoff: tuple = ("uniform", -10, 10),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.db_cutoff = db_cutoff
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        return {"db_cutoff": util.sample_from_dist(self.db_cutoff, state)}
+
+    def _transform(self, signal, db_cutoff: float):
+        return signal.mask_low_magnitudes(db_cutoff)
+
+
+class Smoothing(BaseTransform):
+    """Convolves the signal with a smoothing window.
+
+    Uses :py:func:`audiotools.core.effects.EffectMixin.convolve`.
+
+    Parameters
+    ----------
+    window_type : tuple, optional
+        Type of window to use, by default ("const", "average")
+    window_length : tuple, optional
+        Length of smoothing window, by
+        default ("choice", [8, 16, 32, 64, 128, 256, 512])
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        window_type: tuple = ("const", "average"),
+        window_length: tuple = ("choice", [8, 16, 32, 64, 128, 256, 512]),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(name=name, prob=prob)
+        self.window_type = window_type
+        self.window_length = window_length
+
+    def _instantiate(self, state: RandomState, signal: AudioSignal = None):
+        window_type = util.sample_from_dist(self.window_type, state)
+        window_length = util.sample_from_dist(self.window_length, state)
+        window = signal.get_window(
+            window_type=window_type, window_length=window_length, device="cpu"
+        )
+        return {"window": AudioSignal(window, signal.sample_rate)}
+
+    def _transform(self, signal, window):
+        sscale = signal.audio_data.abs().max(dim=-1, keepdim=True).values
+        sscale[sscale == 0.0] = 1.0
+
+        out = signal.convolve(window)
+
+        oscale = out.audio_data.abs().max(dim=-1, keepdim=True).values
+        oscale[oscale == 0.0] = 1.0
+
+        out = out * (sscale / oscale)
+        return out
+
+
+class TimeNoise(TimeMask):
+    """Similar to :py:func:`audiotools.data.transforms.TimeMask`, but
+    replaces with noise instead of zeros.
+
+    Parameters
+    ----------
+    t_center : tuple, optional
+        Center time in terms of 0.0 and 1.0 (duration of signal),
+        by default ("uniform", 0.0, 1.0)
+    t_width : tuple, optional
+        Width of dropped out portion, by default ("const", 0.025)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        t_center: tuple = ("uniform", 0.0, 1.0),
+        t_width: tuple = ("const", 0.025),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(t_center=t_center, t_width=t_width, name=name, prob=prob)
+
+    def _transform(self, signal, tmin_s: float, tmax_s: float):
+        signal = signal.mask_timesteps(tmin_s=tmin_s, tmax_s=tmax_s, val=0.0)
+        mag, phase = signal.magnitude, signal.phase
+
+        mag_r, phase_r = torch.randn_like(mag), torch.randn_like(phase)
+        mask = (mag == 0.0) * (phase == 0.0)
+
+        mag[mask] = mag_r[mask]
+        phase[mask] = phase_r[mask]
+
+        signal.magnitude = mag
+        signal.phase = phase
+        return signal
+
+
+class FrequencyNoise(FrequencyMask):
+    """Similar to :py:func:`audiotools.data.transforms.FrequencyMask`, but
+    replaces with noise instead of zeros.
+
+    Parameters
+    ----------
+    f_center : tuple, optional
+        Center frequency between 0.0 and 1.0 (Nyquist), by default ("uniform", 0.0, 1.0)
+    f_width : tuple, optional
+        Width of zero'd out band, by default ("const", 0.1)
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        f_center: tuple = ("uniform", 0.0, 1.0),
+        f_width: tuple = ("const", 0.1),
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(f_center=f_center, f_width=f_width, name=name, prob=prob)
+
+    def _transform(self, signal, fmin_hz: float, fmax_hz: float):
+        signal = signal.mask_frequencies(fmin_hz=fmin_hz, fmax_hz=fmax_hz)
+        mag, phase = signal.magnitude, signal.phase
+
+        mag_r, phase_r = torch.randn_like(mag), torch.randn_like(phase)
+        mask = (mag == 0.0) * (phase == 0.0)
+
+        mag[mask] = mag_r[mask]
+        phase[mask] = phase_r[mask]
+
+        signal.magnitude = mag
+        signal.phase = phase
+        return signal
+
+
+class SpectralDenoising(Equalizer):
+    """Applies denoising algorithm detailed in
+    :py:func:`audiotools.ml.layers.spectral_gate.SpectralGate`,
+    using a randomly generated noise signal for denoising.
+
+    Parameters
+    ----------
+    eq_amount : tuple, optional
+        Amount of eq to apply to noise signal, by default ("const", 1.0)
+    denoise_amount : tuple, optional
+        Amount to denoise by, by default ("uniform", 0.8, 1.0)
+    nz_volume : float, optional
+        Volume of noise to denoise with, by default -40
+    n_bands : int, optional
+        Number of bands in equalizer, by default 6
+    n_freq : int, optional
+        Number of frequency bins to smooth by, by default 3
+    n_time : int, optional
+        Number of time bins to smooth by, by default 5
+    name : str, optional
+        Name of this transform, used to identify it in the dictionary
+        produced by ``self.instantiate``, by default None
+    prob : float, optional
+        Probability of applying this transform, by default 1.0
+    """
+
+    def __init__(
+        self,
+        eq_amount: tuple = ("const", 1.0),
+        denoise_amount: tuple = ("uniform", 0.8, 1.0),
+        nz_volume: float = -40,
+        n_bands: int = 6,
+        n_freq: int = 3,
+        n_time: int = 5,
+        name: str = None,
+        prob: float = 1,
+    ):
+        super().__init__(eq_amount=eq_amount, n_bands=n_bands, name=name, prob=prob)
+
+        self.nz_volume = nz_volume
+        self.denoise_amount = denoise_amount
+        self.spectral_gate = ml.layers.SpectralGate(n_freq, n_time)
+
+    def _transform(self, signal, nz, eq, denoise_amount):
+        nz = nz.normalize(self.nz_volume).equalizer(eq)
+        self.spectral_gate = self.spectral_gate.to(signal.device)
+        signal = self.spectral_gate(signal, nz, denoise_amount)
+        return signal
+
+    def _instantiate(self, state: RandomState):
+        kwargs = super()._instantiate(state)
+        kwargs["denoise_amount"] = util.sample_from_dist(self.denoise_amount, state)
+        kwargs["nz"] = AudioSignal(state.randn(22050), 44100)
+        return kwargs

dac-vae/audiotools/metrics/__init__.py

@@ -0,0 +1,6 @@
+"""
+Functions for comparing AudioSignal objects to one another.
+"""  # fmt: skip
+from . import distance
+from . import quality
+from . import spectral

dac-vae/audiotools/metrics/distance.py

@@ -0,0 +1,131 @@
+import torch
+from torch import nn
+
+from .. import AudioSignal
+
+
+class L1Loss(nn.L1Loss):
+    """L1 Loss between AudioSignals. Defaults
+    to comparing ``audio_data``, but any
+    attribute of an AudioSignal can be used.
+
+    Parameters
+    ----------
+    attribute : str, optional
+        Attribute of signal to compare, defaults to ``audio_data``.
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+    """
+
+    def __init__(self, attribute: str = "audio_data", weight: float = 1.0, **kwargs):
+        self.attribute = attribute
+        self.weight = weight
+        super().__init__(**kwargs)
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate AudioSignal
+        y : AudioSignal
+            Reference AudioSignal
+
+        Returns
+        -------
+        torch.Tensor
+            L1 loss between AudioSignal attributes.
+        """
+        if isinstance(x, AudioSignal):
+            x = getattr(x, self.attribute)
+            y = getattr(y, self.attribute)
+        return super().forward(x, y)
+
+
+class SISDRLoss(nn.Module):
+    """
+    Computes the Scale-Invariant Source-to-Distortion Ratio between a batch
+    of estimated and reference audio signals or aligned features.
+
+    Parameters
+    ----------
+    scaling : int, optional
+        Whether to use scale-invariant (True) or
+        signal-to-noise ratio (False), by default True
+    reduction : str, optional
+        How to reduce across the batch (either 'mean',
+        'sum', or none).], by default ' mean'
+    zero_mean : int, optional
+        Zero mean the references and estimates before
+        computing the loss, by default True
+    clip_min : int, optional
+        The minimum possible loss value. Helps network
+        to not focus on making already good examples better, by default None
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+    """
+
+    def __init__(
+        self,
+        scaling: int = True,
+        reduction: str = "mean",
+        zero_mean: int = True,
+        clip_min: int = None,
+        weight: float = 1.0,
+    ):
+        self.scaling = scaling
+        self.reduction = reduction
+        self.zero_mean = zero_mean
+        self.clip_min = clip_min
+        self.weight = weight
+        super().__init__()
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        eps = 1e-8
+        # nb, nc, nt
+        if isinstance(x, AudioSignal):
+            references = x.audio_data
+            estimates = y.audio_data
+        else:
+            references = x
+            estimates = y
+
+        nb = references.shape[0]
+        references = references.reshape(nb, 1, -1).permute(0, 2, 1)
+        estimates = estimates.reshape(nb, 1, -1).permute(0, 2, 1)
+
+        # samples now on axis 1
+        if self.zero_mean:
+            mean_reference = references.mean(dim=1, keepdim=True)
+            mean_estimate = estimates.mean(dim=1, keepdim=True)
+        else:
+            mean_reference = 0
+            mean_estimate = 0
+
+        _references = references - mean_reference
+        _estimates = estimates - mean_estimate
+
+        references_projection = (_references**2).sum(dim=-2) + eps
+        references_on_estimates = (_estimates * _references).sum(dim=-2) + eps
+
+        scale = (
+            (references_on_estimates / references_projection).unsqueeze(1)
+            if self.scaling
+            else 1
+        )
+
+        e_true = scale * _references
+        e_res = _estimates - e_true
+
+        signal = (e_true**2).sum(dim=1)
+        noise = (e_res**2).sum(dim=1)
+        sdr = -10 * torch.log10(signal / noise + eps)
+
+        if self.clip_min is not None:
+            sdr = torch.clamp(sdr, min=self.clip_min)
+
+        if self.reduction == "mean":
+            sdr = sdr.mean()
+        elif self.reduction == "sum":
+            sdr = sdr.sum()
+        return sdr

dac-vae/audiotools/metrics/quality.py

@@ -0,0 +1,159 @@
+import os
+
+import numpy as np
+import torch
+
+from .. import AudioSignal
+
+
+def stoi(
+    estimates: AudioSignal,
+    references: AudioSignal,
+    extended: int = False,
+):
+    """Short term objective intelligibility
+    Computes the STOI (See [1][2]) of a denoised signal compared to a clean
+    signal, The output is expected to have a monotonic relation with the
+    subjective speech-intelligibility, where a higher score denotes better
+    speech intelligibility. Uses pystoi under the hood.
+
+    Parameters
+    ----------
+    estimates : AudioSignal
+        Denoised speech
+    references : AudioSignal
+        Clean original speech
+    extended : int, optional
+        Boolean, whether to use the extended STOI described in [3], by default False
+
+    Returns
+    -------
+    Tensor[float]
+        Short time objective intelligibility measure between clean and
+        denoised speech
+
+    References
+    ----------
+    1.  C.H.Taal, R.C.Hendriks, R.Heusdens, J.Jensen 'A Short-Time
+        Objective Intelligibility Measure for Time-Frequency Weighted Noisy
+        Speech', ICASSP 2010, Texas, Dallas.
+    2.  C.H.Taal, R.C.Hendriks, R.Heusdens, J.Jensen 'An Algorithm for
+        Intelligibility Prediction of Time-Frequency Weighted Noisy Speech',
+        IEEE Transactions on Audio, Speech, and Language Processing, 2011.
+    3.  Jesper Jensen and Cees H. Taal, 'An Algorithm for Predicting the
+        Intelligibility of Speech Masked by Modulated Noise Maskers',
+        IEEE Transactions on Audio, Speech and Language Processing, 2016.
+    """
+    import pystoi
+
+    estimates = estimates.clone().to_mono()
+    references = references.clone().to_mono()
+
+    stois = []
+    for i in range(estimates.batch_size):
+        _stoi = pystoi.stoi(
+            references.audio_data[i, 0].detach().cpu().numpy(),
+            estimates.audio_data[i, 0].detach().cpu().numpy(),
+            references.sample_rate,
+            extended=extended,
+        )
+        stois.append(_stoi)
+    return torch.from_numpy(np.array(stois))
+
+
+def pesq(
+    estimates: AudioSignal,
+    references: AudioSignal,
+    mode: str = "wb",
+    target_sr: float = 16000,
+):
+    """_summary_
+
+    Parameters
+    ----------
+    estimates : AudioSignal
+        Degraded AudioSignal
+    references : AudioSignal
+        Reference AudioSignal
+    mode : str, optional
+        'wb' (wide-band) or 'nb' (narrow-band), by default "wb"
+    target_sr : float, optional
+        Target sample rate, by default 16000
+
+    Returns
+    -------
+    Tensor[float]
+        PESQ score: P.862.2 Prediction (MOS-LQO)
+    """
+    from pesq import pesq as pesq_fn
+
+    estimates = estimates.clone().to_mono().resample(target_sr)
+    references = references.clone().to_mono().resample(target_sr)
+
+    pesqs = []
+    for i in range(estimates.batch_size):
+        _pesq = pesq_fn(
+            estimates.sample_rate,
+            references.audio_data[i, 0].detach().cpu().numpy(),
+            estimates.audio_data[i, 0].detach().cpu().numpy(),
+            mode,
+        )
+        pesqs.append(_pesq)
+    return torch.from_numpy(np.array(pesqs))
+
+
+def visqol(
+    estimates: AudioSignal,
+    references: AudioSignal,
+    mode: str = "audio",
+):  # pragma: no cover
+    """ViSQOL score.
+
+    Parameters
+    ----------
+    estimates : AudioSignal
+        Degraded AudioSignal
+    references : AudioSignal
+        Reference AudioSignal
+    mode : str, optional
+        'audio' or 'speech', by default 'audio'
+
+    Returns
+    -------
+    Tensor[float]
+        ViSQOL score (MOS-LQO)
+    """
+    from visqol import visqol_lib_py
+    from visqol.pb2 import visqol_config_pb2
+    from visqol.pb2 import similarity_result_pb2
+
+    config = visqol_config_pb2.VisqolConfig()
+    if mode == "audio":
+        target_sr = 48000
+        config.options.use_speech_scoring = False
+        svr_model_path = "libsvm_nu_svr_model.txt"
+    elif mode == "speech":
+        target_sr = 16000
+        config.options.use_speech_scoring = True
+        svr_model_path = "lattice_tcditugenmeetpackhref_ls2_nl60_lr12_bs2048_learn.005_ep2400_train1_7_raw.tflite"
+    else:
+        raise ValueError(f"Unrecognized mode: {mode}")
+    config.audio.sample_rate = target_sr
+    config.options.svr_model_path = os.path.join(
+        os.path.dirname(visqol_lib_py.__file__), "model", svr_model_path
+    )
+
+    api = visqol_lib_py.VisqolApi()
+    api.Create(config)
+
+    estimates = estimates.clone().to_mono().resample(target_sr)
+    references = references.clone().to_mono().resample(target_sr)
+
+    visqols = []
+    for i in range(estimates.batch_size):
+        _visqol = api.Measure(
+            references.audio_data[i, 0].detach().cpu().numpy().astype(float),
+            estimates.audio_data[i, 0].detach().cpu().numpy().astype(float),
+        )
+        visqols.append(_visqol.moslqo)
+    return torch.from_numpy(np.array(visqols))

dac-vae/audiotools/metrics/spectral.py

@@ -0,0 +1,247 @@
+import typing
+from typing import List
+
+import numpy as np
+from torch import nn
+
+from .. import AudioSignal
+from .. import STFTParams
+
+
+class MultiScaleSTFTLoss(nn.Module):
+    """Computes the multi-scale STFT loss from [1].
+
+    Parameters
+    ----------
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+
+    References
+    ----------
+
+    1.  Engel, Jesse, Chenjie Gu, and Adam Roberts.
+        "DDSP: Differentiable Digital Signal Processing."
+        International Conference on Learning Representations. 2019.
+    """
+
+    def __init__(
+        self,
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.loss_fn = loss_fn
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.clamp_eps = clamp_eps
+        self.weight = weight
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes multi-scale STFT between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Multi-scale STFT loss.
+        """
+        loss = 0.0
+        for s in self.stft_params:
+            x.stft(s.window_length, s.hop_length, s.window_type)
+            y.stft(s.window_length, s.hop_length, s.window_type)
+            loss += self.log_weight * self.loss_fn(
+                x.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x.magnitude, y.magnitude)
+        return loss
+
+
+class MelSpectrogramLoss(nn.Module):
+    """Compute distance between mel spectrograms. Can be used
+    in a multi-scale way.
+
+    Parameters
+    ----------
+    n_mels : List[int]
+        Number of mels per STFT, by default [150, 80],
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+    """
+
+    def __init__(
+        self,
+        n_mels: List[int] = [150, 80],
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        mel_fmin: List[float] = [0.0, 0.0],
+        mel_fmax: List[float] = [None, None],
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.n_mels = n_mels
+        self.loss_fn = loss_fn
+        self.clamp_eps = clamp_eps
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.weight = weight
+        self.mel_fmin = mel_fmin
+        self.mel_fmax = mel_fmax
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes mel loss between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Mel loss.
+        """
+        loss = 0.0
+        for n_mels, fmin, fmax, s in zip(
+            self.n_mels, self.mel_fmin, self.mel_fmax, self.stft_params
+        ):
+            kwargs = {
+                "window_length": s.window_length,
+                "hop_length": s.hop_length,
+                "window_type": s.window_type,
+            }
+            x_mels = x.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+            y_mels = y.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+
+            loss += self.log_weight * self.loss_fn(
+                x_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x_mels, y_mels)
+        return loss
+
+
+class PhaseLoss(nn.Module):
+    """Difference between phase spectrograms.
+
+    Parameters
+    ----------
+    window_length : int, optional
+        Length of STFT window, by default 2048
+    hop_length : int, optional
+        Hop length of STFT window, by default 512
+    weight : float, optional
+        Weight of loss, by default 1.0
+    """
+
+    def __init__(
+        self, window_length: int = 2048, hop_length: int = 512, weight: float = 1.0
+    ):
+        super().__init__()
+
+        self.weight = weight
+        self.stft_params = STFTParams(window_length, hop_length)
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes phase loss between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Phase loss.
+        """
+        s = self.stft_params
+        x.stft(s.window_length, s.hop_length, s.window_type)
+        y.stft(s.window_length, s.hop_length, s.window_type)
+
+        # Take circular difference
+        diff = x.phase - y.phase
+        diff[diff < -np.pi] += 2 * np.pi
+        diff[diff > np.pi] -= -2 * np.pi
+
+        # Scale true magnitude to weights in [0, 1]
+        x_min, x_max = x.magnitude.min(), x.magnitude.max()
+        weights = (x.magnitude - x_min) / (x_max - x_min)
+
+        # Take weighted mean of all phase errors
+        loss = ((weights * diff) ** 2).mean()
+        return loss

dac-vae/audiotools/ml/__init__.py

@@ -0,0 +1,5 @@
+from . import decorators
+from . import layers
+from .accelerator import Accelerator
+from .experiment import Experiment
+from .layers import BaseModel

dac-vae/audiotools/ml/accelerator.py

@@ -0,0 +1,184 @@
+import os
+import typing
+
+import torch
+import torch.distributed as dist
+from torch.nn.parallel import DataParallel
+from torch.nn.parallel import DistributedDataParallel
+
+from ..data.datasets import ResumableDistributedSampler as DistributedSampler
+from ..data.datasets import ResumableSequentialSampler as SequentialSampler
+
+
+class Accelerator:  # pragma: no cover
+    """This class is used to prepare models and dataloaders for
+    usage with DDP or DP. Use the functions prepare_model, prepare_dataloader to
+    prepare the respective objects. In the case of models, they are moved to
+    the appropriate GPU and SyncBatchNorm is applied to them. In the case of
+    dataloaders, a sampler is created and the dataloader is initialized with
+    that sampler.
+
+    If the world size is 1, prepare_model and prepare_dataloader are
+    no-ops. If the environment variable ``LOCAL_RANK`` is not set, then the
+    script was launched without ``torchrun``, and ``DataParallel``
+    will be used instead of ``DistributedDataParallel`` (not recommended), if
+    the world size (number of GPUs) is greater than 1.
+
+    Parameters
+    ----------
+    amp : bool, optional
+        Whether or not to enable automatic mixed precision, by default False
+    """
+
+    def __init__(self, amp: bool = False):
+        local_rank = os.getenv("LOCAL_RANK", None)
+        self.world_size = torch.cuda.device_count()
+
+        self.use_ddp = self.world_size > 1 and local_rank is not None
+        self.use_dp = self.world_size > 1 and local_rank is None
+        self.device = "cpu" if self.world_size == 0 else "cuda"
+
+        if self.use_ddp:
+            local_rank = int(local_rank)
+            dist.init_process_group(
+                "nccl",
+                init_method="env://",
+                world_size=self.world_size,
+                rank=local_rank,
+            )
+
+        self.local_rank = 0 if local_rank is None else local_rank
+        self.amp = amp
+
+        class DummyScaler:
+            def __init__(self):
+                pass
+
+            def step(self, optimizer):
+                optimizer.step()
+
+            def scale(self, loss):
+                return loss
+
+            def unscale_(self, optimizer):
+                return optimizer
+
+            def update(self):
+                pass
+
+        self.scaler = torch.cuda.amp.GradScaler() if amp else DummyScaler()
+        self.device_ctx = (
+            torch.cuda.device(self.local_rank) if torch.cuda.is_available() else None
+        )
+
+    def __enter__(self):
+        if self.device_ctx is not None:
+            self.device_ctx.__enter__()
+        return self
+
+    def __exit__(self, exc_type, exc_value, traceback):
+        if self.device_ctx is not None:
+            self.device_ctx.__exit__(exc_type, exc_value, traceback)
+
+    def prepare_model(self, model: torch.nn.Module, **kwargs):
+        """Prepares model for DDP or DP. The model is moved to
+        the device of the correct rank.
+
+        Parameters
+        ----------
+        model : torch.nn.Module
+            Model that is converted for DDP or DP.
+
+        Returns
+        -------
+        torch.nn.Module
+            Wrapped model, or original model if DDP and DP are turned off.
+        """
+        model = model.to(self.device)
+        if self.use_ddp:
+            model = torch.nn.SyncBatchNorm.convert_sync_batchnorm(model)
+            model = DistributedDataParallel(
+                model, device_ids=[self.local_rank], **kwargs
+            )
+        elif self.use_dp:
+            model = DataParallel(model, **kwargs)
+        return model
+
+    # Automatic mixed-precision utilities
+    def autocast(self, *args, **kwargs):
+        """Context manager for autocasting. Arguments
+        go to ``torch.cuda.amp.autocast``.
+        """
+        return torch.cuda.amp.autocast(self.amp, *args, **kwargs)
+
+    def backward(self, loss: torch.Tensor):
+        """Backwards pass, after scaling the loss if ``amp`` is
+        enabled.
+
+        Parameters
+        ----------
+        loss : torch.Tensor
+            Loss value.
+        """
+        self.scaler.scale(loss).backward()
+
+    def step(self, optimizer: torch.optim.Optimizer):
+        """Steps the optimizer, using a ``scaler`` if ``amp`` is
+        enabled.
+
+        Parameters
+        ----------
+        optimizer : torch.optim.Optimizer
+            Optimizer to step forward.
+        """
+        self.scaler.step(optimizer)
+
+    def update(self):
+        """Updates the scale factor."""
+        self.scaler.update()
+
+    def prepare_dataloader(
+        self, dataset: typing.Iterable, start_idx: int = None, **kwargs
+    ):
+        """Wraps a dataset with a DataLoader, using the correct sampler if DDP is
+        enabled.
+
+        Parameters
+        ----------
+        dataset : typing.Iterable
+            Dataset to build Dataloader around.
+        start_idx : int, optional
+            Start index of sampler, useful if resuming from some epoch,
+            by default None
+
+        Returns
+        -------
+        _type_
+            _description_
+        """
+
+        if self.use_ddp:
+            sampler = DistributedSampler(
+                dataset,
+                start_idx,
+                num_replicas=self.world_size,
+                rank=self.local_rank,
+            )
+            if "num_workers" in kwargs:
+                kwargs["num_workers"] = max(kwargs["num_workers"] // self.world_size, 1)
+            kwargs["batch_size"] = max(kwargs["batch_size"] // self.world_size, 1)
+        else:
+            sampler = SequentialSampler(dataset, start_idx)
+
+        dataloader = torch.utils.data.DataLoader(dataset, sampler=sampler, **kwargs)
+        return dataloader
+
+    @staticmethod
+    def unwrap(model):
+        """Unwraps the model if it was wrapped in DDP or DP, otherwise
+        just returns the model. Use this to unwrap the model returned by
+        :py:func:`audiotools.ml.accelerator.Accelerator.prepare_model`.
+        """
+        if hasattr(model, "module"):
+            return model.module
+        return model

dac-vae/audiotools/ml/decorators.py

@@ -0,0 +1,441 @@
+import math
+import os
+import time
+from collections import defaultdict
+from functools import wraps
+
+import torch
+import torch.distributed as dist
+from rich import box
+from rich.console import Console
+from rich.console import Group
+from rich.live import Live
+from rich.markdown import Markdown
+from rich.padding import Padding
+from rich.panel import Panel
+from rich.progress import BarColumn
+from rich.progress import Progress
+from rich.progress import SpinnerColumn
+from rich.progress import TimeElapsedColumn
+from rich.progress import TimeRemainingColumn
+from rich.rule import Rule
+from rich.table import Table
+from torch.utils.tensorboard import SummaryWriter
+
+
+# This is here so that the history can be pickled.
+def default_list():
+    return []
+
+
+class Mean:
+    """Keeps track of the running mean, along with the latest
+    value.
+    """
+
+    def __init__(self):
+        self.reset()
+
+    def __call__(self):
+        mean = self.total / max(self.count, 1)
+        return mean
+
+    def reset(self):
+        self.count = 0
+        self.total = 0
+
+    def update(self, val):
+        if math.isfinite(val):
+            self.count += 1
+            self.total += val
+
+
+def when(condition):
+    """Runs a function only when the condition is met. The condition is
+    a function that is run.
+
+    Parameters
+    ----------
+    condition : Callable
+        Function to run to check whether or not to run the decorated
+        function.
+
+    Example
+    -------
+    Checkpoint only runs every 100 iterations, and only if the
+    local rank is 0.
+
+    >>> i = 0
+    >>> rank = 0
+    >>>
+    >>> @when(lambda: i % 100 == 0 and rank == 0)
+    >>> def checkpoint():
+    >>>     print("Saving to /runs/exp1")
+    >>>
+    >>> for i in range(1000):
+    >>>     checkpoint()
+
+    """
+
+    def decorator(fn):
+        @wraps(fn)
+        def decorated(*args, **kwargs):
+            if condition():
+                return fn(*args, **kwargs)
+
+        return decorated
+
+    return decorator
+
+
+def timer(prefix: str = "time"):
+    """Adds execution time to the output dictionary of the decorated
+    function. The function decorated by this must output a dictionary.
+    The key added will follow the form "[prefix]/[name_of_function]"
+
+    Parameters
+    ----------
+    prefix : str, optional
+        The key added will follow the form "[prefix]/[name_of_function]",
+        by default "time".
+    """
+
+    def decorator(fn):
+        @wraps(fn)
+        def decorated(*args, **kwargs):
+            s = time.perf_counter()
+            output = fn(*args, **kwargs)
+            assert isinstance(output, dict)
+            e = time.perf_counter()
+            output[f"{prefix}/{fn.__name__}"] = e - s
+            return output
+
+        return decorated
+
+    return decorator
+
+
+class Tracker:
+    """
+    A tracker class that helps to monitor the progress of training and logging the metrics.
+
+    Attributes
+    ----------
+    metrics : dict
+        A dictionary containing the metrics for each label.
+    history : dict
+        A dictionary containing the history of metrics for each label.
+    writer : SummaryWriter
+        A SummaryWriter object for logging the metrics.
+    rank : int
+        The rank of the current process.
+    step : int
+        The current step of the training.
+    tasks : dict
+        A dictionary containing the progress bars and tables for each label.
+    pbar : Progress
+        A progress bar object for displaying the progress.
+    consoles : list
+        A list of console objects for logging.
+    live : Live
+        A Live object for updating the display live.
+
+    Methods
+    -------
+    print(msg: str)
+        Prints the given message to all consoles.
+    update(label: str, fn_name: str)
+        Updates the progress bar and table for the given label.
+    done(label: str, title: str)
+        Resets the progress bar and table for the given label and prints the final result.
+    track(label: str, length: int, completed: int = 0, op: dist.ReduceOp = dist.ReduceOp.AVG, ddp_active: bool = "LOCAL_RANK" in os.environ)
+        A decorator for tracking the progress and metrics of a function.
+    log(label: str, value_type: str = "value", history: bool = True)
+        A decorator for logging the metrics of a function.
+    is_best(label: str, key: str) -> bool
+        Checks if the latest value of the given key in the label is the best so far.
+    state_dict() -> dict
+        Returns a dictionary containing the state of the tracker.
+    load_state_dict(state_dict: dict) -> Tracker
+        Loads the state of the tracker from the given state dictionary.
+    """
+
+    def __init__(
+        self,
+        writer: SummaryWriter = None,
+        log_file: str = None,
+        rank: int = 0,
+        console_width: int = 100,
+        step: int = 0,
+    ):
+        """
+        Initializes the Tracker object.
+
+        Parameters
+        ----------
+        writer : SummaryWriter, optional
+            A SummaryWriter object for logging the metrics, by default None.
+        log_file : str, optional
+            The path to the log file, by default None.
+        rank : int, optional
+            The rank of the current process, by default 0.
+        console_width : int, optional
+            The width of the console, by default 100.
+        step : int, optional
+            The current step of the training, by default 0.
+        """
+        self.metrics = {}
+        self.history = {}
+        self.writer = writer
+        self.rank = rank
+        self.step = step
+
+        # Create progress bars etc.
+        self.tasks = {}
+        self.pbar = Progress(
+            SpinnerColumn(),
+            "[progress.description]{task.description}",
+            "{task.completed}/{task.total}",
+            BarColumn(),
+            TimeElapsedColumn(),
+            "/",
+            TimeRemainingColumn(),
+        )
+        self.consoles = [Console(width=console_width)]
+        self.live = Live(console=self.consoles[0], refresh_per_second=10)
+        if log_file is not None:
+            self.consoles.append(Console(width=console_width, file=open(log_file, "a")))
+
+    def print(self, msg):
+        """
+        Prints the given message to all consoles.
+
+        Parameters
+        ----------
+        msg : str
+            The message to be printed.
+        """
+        if self.rank == 0:
+            for c in self.consoles:
+                c.log(msg)
+
+    def update(self, label, fn_name):
+        """
+        Updates the progress bar and table for the given label.
+
+        Parameters
+        ----------
+        label : str
+            The label of the progress bar and table to be updated.
+        fn_name : str
+            The name of the function associated with the label.
+        """
+        if self.rank == 0:
+            self.pbar.advance(self.tasks[label]["pbar"])
+
+            # Create table
+            table = Table(title=label, expand=True, box=box.MINIMAL)
+            table.add_column("key", style="cyan")
+            table.add_column("value", style="bright_blue")
+            table.add_column("mean", style="bright_green")
+
+            keys = self.metrics[label]["value"].keys()
+            for k in keys:
+                value = self.metrics[label]["value"][k]
+                mean = self.metrics[label]["mean"][k]()
+                table.add_row(k, f"{value:10.6f}", f"{mean:10.6f}")
+
+            self.tasks[label]["table"] = table
+            tables = [t["table"] for t in self.tasks.values()]
+            group = Group(*tables, self.pbar)
+            self.live.update(
+                Group(
+                    Padding("", (0, 0)),
+                    Rule(f"[italic]{fn_name}()", style="white"),
+                    Padding("", (0, 0)),
+                    Panel.fit(
+                        group, padding=(0, 5), title="[b]Progress", border_style="blue"
+                    ),
+                )
+            )
+
+    def done(self, label: str, title: str):
+        """
+        Resets the progress bar and table for the given label and prints the final result.
+
+        Parameters
+        ----------
+        label : str
+            The label of the progress bar and table to be reset.
+        title : str
+            The title to be displayed when printing the final result.
+        """
+        for label in self.metrics:
+            for v in self.metrics[label]["mean"].values():
+                v.reset()
+
+        if self.rank == 0:
+            self.pbar.reset(self.tasks[label]["pbar"])
+            tables = [t["table"] for t in self.tasks.values()]
+            group = Group(Markdown(f"# {title}"), *tables, self.pbar)
+            self.print(group)
+
+    def track(
+        self,
+        label: str,
+        length: int,
+        completed: int = 0,
+        op: dist.ReduceOp = dist.ReduceOp.AVG,
+        ddp_active: bool = "LOCAL_RANK" in os.environ,
+    ):
+        """
+        A decorator for tracking the progress and metrics of a function.
+
+        Parameters
+        ----------
+        label : str
+            The label to be associated with the progress and metrics.
+        length : int
+            The total number of iterations to be completed.
+        completed : int, optional
+            The number of iterations already completed, by default 0.
+        op : dist.ReduceOp, optional
+            The reduce operation to be used, by default dist.ReduceOp.AVG.
+        ddp_active : bool, optional
+            Whether the DistributedDataParallel is active, by default "LOCAL_RANK" in os.environ.
+        """
+        self.tasks[label] = {
+            "pbar": self.pbar.add_task(
+                f"[white]Iteration ({label})", total=length, completed=completed
+            ),
+            "table": Table(),
+        }
+        self.metrics[label] = {
+            "value": defaultdict(),
+            "mean": defaultdict(lambda: Mean()),
+        }
+
+        def decorator(fn):
+            @wraps(fn)
+            def decorated(*args, **kwargs):
+                output = fn(*args, **kwargs)
+                if not isinstance(output, dict):
+                    self.update(label, fn.__name__)
+                    return output
+                # Collect across all DDP processes
+                scalar_keys = []
+                for k, v in output.items():
+                    if isinstance(v, (int, float)):
+                        v = torch.tensor([v])
+                    if not torch.is_tensor(v):
+                        continue
+                    if ddp_active and v.is_cuda:  # pragma: no cover
+                        dist.all_reduce(v, op=op)
+                    output[k] = v.detach()
+                    if torch.numel(v) == 1:
+                        scalar_keys.append(k)
+                        output[k] = v.item()
+
+                # Save the outputs to tracker
+                for k, v in output.items():
+                    if k not in scalar_keys:
+                        continue
+                    self.metrics[label]["value"][k] = v
+                    # Update the running mean
+                    self.metrics[label]["mean"][k].update(v)
+
+                self.update(label, fn.__name__)
+                return output
+
+            return decorated
+
+        return decorator
+
+    def log(self, label: str, value_type: str = "value", history: bool = True):
+        """
+        A decorator for logging the metrics of a function.
+
+        Parameters
+        ----------
+        label : str
+            The label to be associated with the logging.
+        value_type : str, optional
+            The type of value to be logged, by default "value".
+        history : bool, optional
+            Whether to save the history of the metrics, by default True.
+        """
+        assert value_type in ["mean", "value"]
+        if history:
+            if label not in self.history:
+                self.history[label] = defaultdict(default_list)
+
+        def decorator(fn):
+            @wraps(fn)
+            def decorated(*args, **kwargs):
+                output = fn(*args, **kwargs)
+                if self.rank == 0:
+                    nonlocal value_type, label
+                    metrics = self.metrics[label][value_type]
+                    for k, v in metrics.items():
+                        v = v() if isinstance(v, Mean) else v
+                        if self.writer is not None:
+                            # self.writer.add_scalar(f"{k}/{label}", v, self.step)
+                            self.writer.log_metric(f"{k}_{label}", v, step=self.step)
+                        if label in self.history:
+                            self.history[label][k].append(v)
+
+                    if label in self.history:
+                        self.history[label]["step"].append(self.step)
+
+                return output
+
+            return decorated
+
+        return decorator
+
+    def is_best(self, label, key):
+        """
+        Checks if the latest value of the given key in the label is the best so far.
+
+        Parameters
+        ----------
+        label : str
+            The label of the metrics to be checked.
+        key : str
+            The key of the metric to be checked.
+
+        Returns
+        -------
+        bool
+            True if the latest value is the best so far, otherwise False.
+        """
+        return self.history[label][key][-1] == min(self.history[label][key])
+
+    def state_dict(self):
+        """
+        Returns a dictionary containing the state of the tracker.
+
+        Returns
+        -------
+        dict
+            A dictionary containing the history and step of the tracker.
+        """
+        return {"history": self.history, "step": self.step}
+
+    def load_state_dict(self, state_dict):
+        """
+        Loads the state of the tracker from the given state dictionary.
+
+        Parameters
+        ----------
+        state_dict : dict
+            A dictionary containing the history and step of the tracker.
+
+        Returns
+        -------
+        Tracker
+            The tracker object with the loaded state.
+        """
+        self.history = state_dict["history"]
+        self.step = state_dict["step"]
+        return self