Add monkey patched fairseq package to run on python 3.11 (what is needed for our use of RVC at least)

modules/voice_conversion/fairseq/data/audio/__init__.py

@@ -0,0 +1,93 @@
+from abc import ABC, abstractmethod
+from typing import Dict, Optional
+import importlib
+import os
+import numpy as np
+
+
+class AudioTransform(ABC):
+    @classmethod
+    @abstractmethod
+    def from_config_dict(cls, config: Optional[Dict] = None):
+        pass
+
+
+class CompositeAudioTransform(AudioTransform):
+    def _from_config_dict(
+        cls,
+        transform_type,
+        get_audio_transform,
+        composite_cls,
+        config=None,
+        return_empty=False,
+    ):
+        _config = {} if config is None else config
+        _transforms = _config.get(f"{transform_type}_transforms")
+
+        if _transforms is None:
+            if return_empty:
+                _transforms = []
+            else:
+                return None
+
+        transforms = [
+            get_audio_transform(_t).from_config_dict(_config.get(_t))
+            for _t in _transforms
+        ]
+        return composite_cls(transforms)
+
+    def __init__(self, transforms):
+        self.transforms = [t for t in transforms if t is not None]
+
+    def __call__(self, x):
+        for t in self.transforms:
+            x = t(x)
+        return x
+
+    def __repr__(self):
+        format_string = (
+            [self.__class__.__name__ + "("]
+            + [f"    {t.__repr__()}" for t in self.transforms]
+            + [")"]
+        )
+        return "\n".join(format_string)
+
+
+def register_audio_transform(name, cls_type, registry, class_names):
+    def register_audio_transform_cls(cls):
+        if name in registry:
+            raise ValueError(f"Cannot register duplicate transform ({name})")
+        if not issubclass(cls, cls_type):
+            raise ValueError(
+                f"Transform ({name}: {cls.__name__}) must extend "
+                f"{cls_type.__name__}"
+            )
+        if cls.__name__ in class_names:
+            raise ValueError(
+                f"Cannot register audio transform with duplicate "
+                f"class name ({cls.__name__})"
+            )
+        registry[name] = cls
+        class_names.add(cls.__name__)
+        return cls
+
+    return register_audio_transform_cls
+
+
+def import_transforms(transforms_dir, transform_type):
+    for file in os.listdir(transforms_dir):
+        path = os.path.join(transforms_dir, file)
+        if (
+            not file.startswith("_")
+            and not file.startswith(".")
+            and (file.endswith(".py") or os.path.isdir(path))
+        ):
+            name = file[: file.find(".py")] if file.endswith(".py") else file
+            importlib.import_module(
+                f"fairseq.data.audio.{transform_type}_transforms." + name
+            )
+
+
+# Utility fn for uniform numbers in transforms
+def rand_uniform(a, b):
+    return np.random.uniform() * (b - a) + a

modules/voice_conversion/fairseq/data/audio/audio_utils.py

@@ -0,0 +1,389 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+
+import mmap
+from pathlib import Path
+import io
+from typing import BinaryIO, List, Optional, Tuple, Union
+
+import numpy as np
+import torch
+import torch.nn.functional as F
+
+from fairseq.data.audio.waveform_transforms import CompositeAudioWaveformTransform
+
+SF_AUDIO_FILE_EXTENSIONS = {".wav", ".flac", ".ogg"}
+FEATURE_OR_SF_AUDIO_FILE_EXTENSIONS = {".npy", ".wav", ".flac", ".ogg"}
+
+
+def convert_waveform(
+    waveform: Union[np.ndarray, torch.Tensor],
+    sample_rate: int,
+    normalize_volume: bool = False,
+    to_mono: bool = False,
+    to_sample_rate: Optional[int] = None,
+) -> Tuple[Union[np.ndarray, torch.Tensor], int]:
+    """convert a waveform:
+    - to a target sample rate
+    - from multi-channel to mono channel
+    - volume normalization
+
+    Args:
+        waveform (numpy.ndarray or torch.Tensor): 2D original waveform
+            (channels x length)
+        sample_rate (int): original sample rate
+        normalize_volume (bool): perform volume normalization
+        to_mono (bool): convert to mono channel if having multiple channels
+        to_sample_rate (Optional[int]): target sample rate
+    Returns:
+        waveform (numpy.ndarray): converted 2D waveform (channels x length)
+        sample_rate (float): target sample rate
+    """
+    try:
+        import torchaudio.sox_effects as ta_sox
+    except ImportError:
+        raise ImportError("Please install torchaudio: pip install torchaudio")
+
+    effects = []
+    if normalize_volume:
+        effects.append(["gain", "-n"])
+    if to_sample_rate is not None and to_sample_rate != sample_rate:
+        effects.append(["rate", f"{to_sample_rate}"])
+    if to_mono and waveform.shape[0] > 1:
+        effects.append(["channels", "1"])
+    if len(effects) > 0:
+        is_np_input = isinstance(waveform, np.ndarray)
+        _waveform = torch.from_numpy(waveform) if is_np_input else waveform
+        converted, converted_sample_rate = ta_sox.apply_effects_tensor(
+            _waveform, sample_rate, effects
+        )
+        if is_np_input:
+            converted = converted.numpy()
+        return converted, converted_sample_rate
+    return waveform, sample_rate
+
+
+def get_waveform(
+    path_or_fp: Union[str, BinaryIO],
+    normalization: bool = True,
+    mono: bool = True,
+    frames: int = -1,
+    start: int = 0,
+    always_2d: bool = True,
+    output_sample_rate: Optional[int] = None,
+    normalize_volume: bool = False,
+    waveform_transforms: Optional[CompositeAudioWaveformTransform] = None,
+) -> Tuple[np.ndarray, int]:
+    """Get the waveform and sample rate of a 16-bit WAV/FLAC/OGG Vorbis audio.
+
+    Args:
+        path_or_fp (str or BinaryIO): the path or file-like object
+        normalization (bool): normalize values to [-1, 1] (Default: True)
+        mono (bool): convert multi-channel audio to mono-channel one
+        frames (int): the number of frames to read. (-1 for reading all)
+        start (int): Where to start reading. A negative value counts from the end.
+        always_2d (bool): always return 2D array even for mono-channel audios
+        output_sample_rate (Optional[int]): output sample rate
+        normalize_volume (bool): normalize volume
+    Returns:
+        waveform (numpy.ndarray): 1D or 2D waveform (channels x length)
+        sample_rate (float): sample rate
+    """
+    if isinstance(path_or_fp, str):
+        ext = Path(path_or_fp).suffix
+        if ext not in SF_AUDIO_FILE_EXTENSIONS:
+            raise ValueError(f"Unsupported audio format: {ext}")
+
+    try:
+        import soundfile as sf
+    except ImportError:
+        raise ImportError("Please install soundfile: pip install soundfile")
+
+    waveform, sample_rate = sf.read(
+        path_or_fp, dtype="float32", always_2d=True, frames=frames, start=start
+    )
+    waveform = waveform.T  # T x C -> C x T
+    waveform, sample_rate = convert_waveform(
+        waveform,
+        sample_rate,
+        normalize_volume=normalize_volume,
+        to_mono=mono,
+        to_sample_rate=output_sample_rate,
+    )
+
+    if not normalization:
+        waveform *= 2**15  # denormalized to 16-bit signed integers
+
+    if waveform_transforms is not None:
+        waveform, sample_rate = waveform_transforms(waveform, sample_rate)
+
+    if not always_2d:
+        waveform = waveform.squeeze(axis=0)
+
+    return waveform, sample_rate
+
+
+def get_features_from_npy_or_audio(path, waveform_transforms=None):
+    ext = Path(path).suffix
+    if ext not in FEATURE_OR_SF_AUDIO_FILE_EXTENSIONS:
+        raise ValueError(f'Unsupported file format for "{path}"')
+    return (
+        np.load(path)
+        if ext == ".npy"
+        else get_fbank(path, waveform_transforms=waveform_transforms)
+    )
+
+
+def get_features_or_waveform_from_stored_zip(
+    path,
+    byte_offset,
+    byte_size,
+    need_waveform=False,
+    use_sample_rate=None,
+    waveform_transforms=None,
+):
+    assert path.endswith(".zip")
+    data = read_from_stored_zip(path, byte_offset, byte_size)
+    f = io.BytesIO(data)
+    if is_npy_data(data):
+        features_or_waveform = np.load(f)
+    elif is_sf_audio_data(data):
+        features_or_waveform = (
+            get_waveform(
+                f,
+                always_2d=False,
+                output_sample_rate=use_sample_rate,
+                waveform_transforms=waveform_transforms,
+            )[0]
+            if need_waveform
+            else get_fbank(f, waveform_transforms=waveform_transforms)
+        )
+    else:
+        raise ValueError(f'Unknown file format for "{path}"')
+    return features_or_waveform
+
+
+def get_features_or_waveform(
+    path: str, need_waveform=False, use_sample_rate=None, waveform_transforms=None
+):
+    """Get speech features from .npy file or waveform from .wav/.flac file.
+    The file may be inside an uncompressed ZIP file and is accessed via byte
+    offset and length.
+
+    Args:
+        path (str): File path in the format of "<.npy/.wav/.flac path>" or
+        "<zip path>:<byte offset>:<byte length>".
+        need_waveform (bool): return waveform instead of features.
+        use_sample_rate (int): change sample rate for the input wave file
+
+    Returns:
+        features_or_waveform (numpy.ndarray): speech features or waveform.
+    """
+    _path, slice_ptr = parse_path(path)
+    if len(slice_ptr) == 0:
+        if need_waveform:
+            return get_waveform(
+                _path,
+                always_2d=False,
+                output_sample_rate=use_sample_rate,
+                waveform_transforms=waveform_transforms,
+            )[0]
+        return get_features_from_npy_or_audio(
+            _path, waveform_transforms=waveform_transforms
+        )
+    elif len(slice_ptr) == 2:
+        features_or_waveform = get_features_or_waveform_from_stored_zip(
+            _path,
+            slice_ptr[0],
+            slice_ptr[1],
+            need_waveform=need_waveform,
+            use_sample_rate=use_sample_rate,
+            waveform_transforms=waveform_transforms,
+        )
+    else:
+        raise ValueError(f"Invalid path: {path}")
+
+    return features_or_waveform
+
+
+def _get_kaldi_fbank(
+    waveform: np.ndarray, sample_rate: int, n_bins=80
+) -> Optional[np.ndarray]:
+    """Get mel-filter bank features via PyKaldi."""
+    try:
+        from kaldi.feat.fbank import Fbank, FbankOptions
+        from kaldi.feat.mel import MelBanksOptions
+        from kaldi.feat.window import FrameExtractionOptions
+        from kaldi.matrix import Vector
+
+        mel_opts = MelBanksOptions()
+        mel_opts.num_bins = n_bins
+        frame_opts = FrameExtractionOptions()
+        frame_opts.samp_freq = sample_rate
+        opts = FbankOptions()
+        opts.mel_opts = mel_opts
+        opts.frame_opts = frame_opts
+        fbank = Fbank(opts=opts)
+        features = fbank.compute(Vector(waveform.squeeze()), 1.0).numpy()
+        return features
+    except ImportError:
+        return None
+
+
+def _get_torchaudio_fbank(
+    waveform: np.ndarray, sample_rate, n_bins=80
+) -> Optional[np.ndarray]:
+    """Get mel-filter bank features via TorchAudio."""
+    try:
+        import torchaudio.compliance.kaldi as ta_kaldi
+
+        waveform = torch.from_numpy(waveform)
+        features = ta_kaldi.fbank(
+            waveform, num_mel_bins=n_bins, sample_frequency=sample_rate
+        )
+        return features.numpy()
+    except ImportError:
+        return None
+
+
+def get_fbank(
+    path_or_fp: Union[str, BinaryIO], n_bins=80, waveform_transforms=None
+) -> np.ndarray:
+    """Get mel-filter bank features via PyKaldi or TorchAudio. Prefer PyKaldi
+    (faster CPP implementation) to TorchAudio (Python implementation). Note that
+    Kaldi/TorchAudio requires 16-bit signed integers as inputs and hence the
+    waveform should not be normalized."""
+    waveform, sample_rate = get_waveform(
+        path_or_fp, normalization=False, waveform_transforms=waveform_transforms
+    )
+
+    features = _get_kaldi_fbank(waveform, sample_rate, n_bins)
+    if features is None:
+        features = _get_torchaudio_fbank(waveform, sample_rate, n_bins)
+    if features is None:
+        raise ImportError(
+            "Please install pyKaldi or torchaudio to enable "
+            "online filterbank feature extraction"
+        )
+
+    return features
+
+
+def is_npy_data(data: bytes) -> bool:
+    return data[0] == 147 and data[1] == 78
+
+
+def is_sf_audio_data(data: bytes) -> bool:
+    is_wav = data[0] == 82 and data[1] == 73 and data[2] == 70
+    is_flac = data[0] == 102 and data[1] == 76 and data[2] == 97
+    is_ogg = data[0] == 79 and data[1] == 103 and data[2] == 103
+    return is_wav or is_flac or is_ogg
+
+
+def mmap_read(path: str, offset: int, length: int) -> bytes:
+    with open(path, "rb") as f:
+        with mmap.mmap(f.fileno(), length=0, access=mmap.ACCESS_READ) as mmap_o:
+            data = mmap_o[offset : offset + length]
+    return data
+
+
+def read_from_stored_zip(zip_path: str, offset: int, length: int) -> bytes:
+    return mmap_read(zip_path, offset, length)
+
+
+def parse_path(path: str) -> Tuple[str, List[int]]:
+    """Parse data path which is either a path to
+    1. a .npy/.wav/.flac/.ogg file
+    2. a stored ZIP file with slicing info: "[zip_path]:[offset]:[length]"
+
+      Args:
+          path (str): the data path to parse
+
+      Returns:
+          file_path (str): the file path
+          slice_ptr (list of int): empty in case 1;
+            byte offset and length for the slice in case 2
+    """
+
+    if Path(path).suffix in FEATURE_OR_SF_AUDIO_FILE_EXTENSIONS:
+        _path, slice_ptr = path, []
+    else:
+        _path, *slice_ptr = path.split(":")
+        if not Path(_path).is_file():
+            raise FileNotFoundError(f"File not found: {_path}")
+    assert len(slice_ptr) in {0, 2}, f"Invalid path: {path}"
+    slice_ptr = [int(i) for i in slice_ptr]
+    return _path, slice_ptr
+
+
+def get_window(window_fn: callable, n_fft: int, win_length: int) -> torch.Tensor:
+    padding = n_fft - win_length
+    assert padding >= 0
+    return F.pad(window_fn(win_length), (padding // 2, padding - padding // 2))
+
+
+def get_fourier_basis(n_fft: int) -> torch.Tensor:
+    basis = np.fft.fft(np.eye(n_fft))
+    basis = np.vstack(
+        [np.real(basis[: n_fft // 2 + 1, :]), np.imag(basis[: n_fft // 2 + 1, :])]
+    )
+    return torch.from_numpy(basis).float()
+
+
+def get_mel_filters(
+    sample_rate: int, n_fft: int, n_mels: int, f_min: float, f_max: float
+) -> torch.Tensor:
+    try:
+        import librosa
+    except ImportError:
+        raise ImportError("Please install librosa: pip install librosa")
+    basis = librosa.filters.mel(sample_rate, n_fft, n_mels, f_min, f_max)
+    return torch.from_numpy(basis).float()
+
+
+class TTSSpectrogram(torch.nn.Module):
+    def __init__(
+        self,
+        n_fft: int,
+        win_length: int,
+        hop_length: int,
+        window_fn: callable = torch.hann_window,
+        return_phase: bool = False,
+    ) -> None:
+        super(TTSSpectrogram, self).__init__()
+        self.n_fft = n_fft
+        self.hop_length = hop_length
+        self.return_phase = return_phase
+
+        basis = get_fourier_basis(n_fft).unsqueeze(1)
+        basis *= get_window(window_fn, n_fft, win_length)
+        self.register_buffer("basis", basis)
+
+    def forward(
+        self, waveform: torch.Tensor
+    ) -> Union[torch.Tensor, Tuple[torch.Tensor, torch.Tensor]]:
+        padding = (self.n_fft // 2, self.n_fft // 2)
+        x = F.pad(waveform.unsqueeze(1), padding, mode="reflect")
+        x = F.conv1d(x, self.basis, stride=self.hop_length)
+        real_part = x[:, : self.n_fft // 2 + 1, :]
+        imag_part = x[:, self.n_fft // 2 + 1 :, :]
+        magnitude = torch.sqrt(real_part**2 + imag_part**2)
+        if self.return_phase:
+            phase = torch.atan2(imag_part, real_part)
+            return magnitude, phase
+        return magnitude
+
+
+class TTSMelScale(torch.nn.Module):
+    def __init__(
+        self, n_mels: int, sample_rate: int, f_min: float, f_max: float, n_stft: int
+    ) -> None:
+        super(TTSMelScale, self).__init__()
+        basis = get_mel_filters(sample_rate, (n_stft - 1) * 2, n_mels, f_min, f_max)
+        self.register_buffer("basis", basis)
+
+    def forward(self, specgram: torch.Tensor) -> torch.Tensor:
+        return torch.matmul(self.basis, specgram)

modules/voice_conversion/fairseq/data/audio/data_cfg.py

@@ -0,0 +1,387 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+import logging
+from argparse import Namespace
+from copy import deepcopy
+from pathlib import Path
+from typing import Dict, Optional
+
+from fairseq.data import Dictionary
+
+logger = logging.getLogger(__name__)
+
+
+def get_config_from_yaml(yaml_path: Path):
+    try:
+        import yaml
+    except ImportError:
+        print("Please install PyYAML: pip install PyYAML")
+    config = {}
+    if yaml_path.is_file():
+        try:
+            with open(yaml_path) as f:
+                config = yaml.load(f, Loader=yaml.FullLoader)
+        except Exception as e:
+            raise Exception(f"Failed to load config from {yaml_path.as_posix()}: {e}")
+    else:
+        raise FileNotFoundError(f"{yaml_path.as_posix()} not found")
+
+    return config
+
+
+class S2TDataConfig(object):
+    """Wrapper class for data config YAML"""
+
+    def __init__(self, yaml_path: Path):
+        self.config = get_config_from_yaml(yaml_path)
+        self.root = yaml_path.parent
+
+    def _auto_convert_to_abs_path(self, x):
+        if isinstance(x, str):
+            if not Path(x).exists() and (self.root / x).exists():
+                return (self.root / x).as_posix()
+        elif isinstance(x, dict):
+            return {k: self._auto_convert_to_abs_path(v) for k, v in x.items()}
+        return x
+
+    @property
+    def vocab_filename(self):
+        """fairseq vocabulary file under data root"""
+        return self.config.get("vocab_filename", "dict.txt")
+
+    @property
+    def speaker_set_filename(self):
+        """speaker set file under data root"""
+        return self.config.get("speaker_set_filename", None)
+
+    @property
+    def shuffle(self) -> bool:
+        """Shuffle dataset samples before batching"""
+        return self.config.get("shuffle", False)
+
+    @property
+    def pre_tokenizer(self) -> Dict:
+        """Pre-tokenizer to apply before subword tokenization. Returning
+        a dictionary with `tokenizer` providing the tokenizer name and
+        the other items providing the tokenizer-specific arguments.
+        Tokenizers are defined in `fairseq.data.encoders.*`"""
+        tokenizer = self.config.get("pre_tokenizer", {"tokenizer": None})
+        return self._auto_convert_to_abs_path(tokenizer)
+
+    @property
+    def bpe_tokenizer(self) -> Dict:
+        """Subword tokenizer to apply after pre-tokenization. Returning
+        a dictionary with `bpe` providing the tokenizer name and
+        the other items providing the tokenizer-specific arguments.
+        Tokenizers are defined in `fairseq.data.encoders.*`"""
+        tokenizer = self.config.get("bpe_tokenizer", {"bpe": None})
+        return self._auto_convert_to_abs_path(tokenizer)
+
+    @property
+    def prepend_tgt_lang_tag(self) -> bool:
+        """Prepend target lang ID token as the target BOS (e.g. for to-many
+        multilingual setting). During inference, this requires `--prefix-size 1`
+        to force BOS to be lang ID token."""
+        return self.config.get("prepend_tgt_lang_tag", False)
+
+    @property
+    def prepend_bos_and_append_tgt_lang_tag(self) -> bool:
+        """Prepend BOS and append target lang ID token to the target (e.g. mBART with language token pretraining)."""
+        return self.config.get("prepend_bos_and_append_tgt_lang_tag", False)
+
+    @property
+    def input_feat_per_channel(self):
+        """The dimension of input features (per audio channel)"""
+        return self.config.get("input_feat_per_channel", 80)
+
+    @property
+    def input_channels(self):
+        """The number of channels in the input audio"""
+        return self.config.get("input_channels", 1)
+
+    @property
+    def sample_rate(self):
+        return self.config.get("sample_rate", 16_000)
+
+    @property
+    def sampling_alpha(self):
+        """Hyper-parameter alpha = 1/T for temperature-based resampling.
+        (alpha = 1 for no resampling)"""
+        return self.config.get("sampling_alpha", 1.0)
+
+    @property
+    def use_audio_input(self):
+        """Needed by the dataset loader to see if the model requires
+        raw audio as inputs."""
+        return self.config.get("use_audio_input", False)
+
+    def standardize_audio(self) -> bool:
+        return self.use_audio_input and self.config.get("standardize_audio", False)
+
+    @property
+    def use_sample_rate(self):
+        """Needed by the dataset loader to see if the model requires
+        raw audio with specific sample rate as inputs."""
+        return self.config.get("use_sample_rate", 16000)
+
+    @property
+    def audio_root(self):
+        """Audio paths in the manifest TSV can be relative and this provides
+        the root path. Set this to empty string when using absolute paths."""
+        return self.config.get("audio_root", "")
+
+    def get_transforms(self, transform_type, split, is_train):
+        """Split-specific feature transforms. Allowing train set
+        wildcard `_train`, evaluation set wildcard `_eval` and general
+        wildcard `*` for matching."""
+        from copy import deepcopy
+
+        cfg = deepcopy(self.config)
+        _cur = cfg.get(f"{transform_type}transforms", {})
+        cur = _cur.get(split)
+        cur = _cur.get("_train") if cur is None and is_train else cur
+        cur = _cur.get("_eval") if cur is None and not is_train else cur
+        cur = _cur.get("*") if cur is None else cur
+        return cur
+
+    def get_feature_transforms(self, split, is_train):
+        cfg = deepcopy(self.config)
+        # TODO: deprecate transforms
+        cur = self.get_transforms("", split, is_train)
+        if cur is not None:
+            logger.warning(
+                "Auto converting transforms into feature_transforms, "
+                "but transforms will be deprecated in the future. Please "
+                "update this in the config."
+            )
+            ft_transforms = self.get_transforms("feature_", split, is_train)
+            if ft_transforms:
+                cur.extend(ft_transforms)
+        else:
+            cur = self.get_transforms("feature_", split, is_train)
+        cfg["feature_transforms"] = cur
+        return cfg
+
+    def get_waveform_transforms(self, split, is_train):
+        cfg = deepcopy(self.config)
+        cfg["waveform_transforms"] = self.get_transforms("waveform_", split, is_train)
+        return cfg
+
+    def get_dataset_transforms(self, split, is_train):
+        cfg = deepcopy(self.config)
+        cfg["dataset_transforms"] = self.get_transforms("dataset_", split, is_train)
+        return cfg
+
+    @property
+    def global_cmvn_stats_npz(self) -> Optional[str]:
+        path = self.config.get("global_cmvn", {}).get("stats_npz_path", None)
+        return self._auto_convert_to_abs_path(path)
+
+    @property
+    def vocoder(self) -> Dict[str, str]:
+        vocoder = self.config.get("vocoder", {"type": "griffin_lim"})
+        return self._auto_convert_to_abs_path(vocoder)
+
+    @property
+    def hub(self) -> Dict[str, str]:
+        return self.config.get("hub", {})
+
+
+class S2SDataConfig(S2TDataConfig):
+    """Wrapper class for data config YAML"""
+
+    @property
+    def vocab_filename(self):
+        """fairseq vocabulary file under data root"""
+        return self.config.get("vocab_filename", None)
+
+    @property
+    def pre_tokenizer(self) -> Dict:
+        return None
+
+    @property
+    def bpe_tokenizer(self) -> Dict:
+        return None
+
+    @property
+    def input_transformed_channels(self):
+        """The number of channels in the audio after feature transforms"""
+        # TODO: move this into individual transforms
+        # TODO: deprecate transforms
+        _cur = self.config.get("transforms", {})
+        ft_transforms = self.config.get("feature_transforms", {})
+        if _cur and ft_transforms:
+            _cur.update(ft_transforms)
+        else:
+            _cur = self.config.get("feature_transforms", {})
+        cur = _cur.get("_train", [])
+
+        _channels = self.input_channels
+        if "delta_deltas" in cur:
+            _channels *= 3
+
+        return _channels
+
+    @property
+    def output_sample_rate(self):
+        """The audio sample rate of output target speech"""
+        return self.config.get("output_sample_rate", 22050)
+
+    @property
+    def target_speaker_embed(self):
+        """Target speaker embedding file (one line per target audio sample)"""
+        return self.config.get("target_speaker_embed", None)
+
+    @property
+    def prepend_tgt_lang_tag_as_bos(self) -> bool:
+        """Prepend target lang ID token as the target BOS."""
+        return self.config.get("prepend_tgt_lang_tag_as_bos", False)
+
+
+class MultitaskConfig(object):
+    """Wrapper class for data config YAML"""
+
+    def __init__(self, yaml_path: Path):
+        config = get_config_from_yaml(yaml_path)
+        self.config = {}
+        for k, v in config.items():
+            self.config[k] = SingleTaskConfig(k, v)
+
+    def get_all_tasks(self):
+        return self.config
+
+    def get_single_task(self, name):
+        assert name in self.config, f"multitask '{name}' does not exist!"
+        return self.config[name]
+
+    @property
+    def first_pass_decoder_task_index(self):
+        """Return the task index of the first-pass text decoder.
+        If there are multiple 'is_first_pass_decoder: True' in the config file,
+            the last task is used for the first-pass decoder.
+        If there is no 'is_first_pass_decoder: True' in the config file,
+            the last task whose task_name includes 'target' and decoder_type is not ctc.
+        """
+        idx = -1
+        for i, (k, v) in enumerate(self.config.items()):
+            if v.is_first_pass_decoder:
+                idx = i
+        if idx < 0:
+            for i, (k, v) in enumerate(self.config.items()):
+                if k.startswith("target") and v.decoder_type == "transformer":
+                    idx = i
+        return idx
+
+
+class SingleTaskConfig(object):
+    def __init__(self, name, config):
+        self.task_name = name
+        self.config = config
+        dict_path = config.get("dict", "")
+        self.tgt_dict = Dictionary.load(dict_path) if Path(dict_path).exists() else None
+
+    @property
+    def data(self):
+        return self.config.get("data", "")
+
+    @property
+    def decoder_type(self):
+        return self.config.get("decoder_type", "transformer")
+
+    @property
+    def decoder_args(self):
+        """Decoder arch related args"""
+        args = self.config.get("decoder_args", {})
+        return Namespace(**args)
+
+    @property
+    def criterion_cfg(self):
+        """cfg for the multitask criterion"""
+        if self.decoder_type == "ctc":
+            from fairseq.criterions.ctc import CtcCriterionConfig
+
+            cfg = CtcCriterionConfig
+            cfg.zero_infinity = self.config.get("zero_infinity", True)
+        else:
+            from fairseq.criterions.label_smoothed_cross_entropy import (
+                LabelSmoothedCrossEntropyCriterionConfig,
+            )
+
+            cfg = LabelSmoothedCrossEntropyCriterionConfig
+            cfg.label_smoothing = self.config.get("label_smoothing", 0.2)
+        return cfg
+
+    @property
+    def input_from(self):
+        """Condition on encoder/decoder of the main model"""
+        return "decoder" if "decoder_layer" in self.config else "encoder"
+
+    @property
+    def input_layer(self):
+        if self.input_from == "decoder":
+            return self.config["decoder_layer"] - 1
+        else:
+            # default using the output from the last encoder layer (-1)
+            return self.config.get("encoder_layer", 0) - 1
+
+    @property
+    def loss_weight_schedule(self):
+        return (
+            "decay"
+            if "loss_weight_max" in self.config
+            and "loss_weight_decay_steps" in self.config
+            else "fixed"
+        )
+
+    def get_loss_weight(self, num_updates):
+        if self.loss_weight_schedule == "fixed":
+            weight = self.config.get("loss_weight", 1.0)
+        else:  # "decay"
+            assert (
+                self.config.get("loss_weight_decay_steps", 0) > 0
+            ), "loss_weight_decay_steps must be greater than 0 for a decay schedule"
+            loss_weight_min = self.config.get("loss_weight_min", 0.0001)
+            loss_weight_decay_stepsize = (
+                self.config["loss_weight_max"] - loss_weight_min
+            ) / self.config["loss_weight_decay_steps"]
+            weight = max(
+                self.config["loss_weight_max"]
+                - loss_weight_decay_stepsize * num_updates,
+                loss_weight_min,
+            )
+        return weight
+
+    @property
+    def prepend_bos_and_append_tgt_lang_tag(self) -> bool:
+        """Prepend BOS and append target lang ID token to the target (e.g. mBART with language token pretraining)."""
+        return self.config.get("prepend_bos_and_append_tgt_lang_tag", False)
+
+    @property
+    def eos_token(self):
+        """EOS token during generation"""
+        return self.config.get("eos_token", "<eos>")
+
+    @property
+    def rdrop_alpha(self):
+        return self.config.get("rdrop_alpha", 0.0)
+
+    @property
+    def is_first_pass_decoder(self):
+        flag = self.config.get("is_first_pass_decoder", False)
+        if flag:
+            if self.decoder_type == "ctc":
+                raise ValueError(
+                    "First-pass decoder in the multi-decoder model must not be CTC."
+                )
+            if "target" not in self.task_name:
+                raise Warning(
+                    'The name of the first-pass decoder does not include "target".'
+                )
+        return flag
+
+    @property
+    def get_lang_tag_mapping(self):
+        return self.config.get("lang_tag_mapping", {})

modules/voice_conversion/fairseq/data/audio/dataset_transforms/__init__.py

@@ -0,0 +1,53 @@
+import os
+from fairseq.data.audio import (
+    AudioTransform,
+    CompositeAudioTransform,
+    import_transforms,
+    register_audio_transform,
+)
+
+
+class AudioDatasetTransform(AudioTransform):
+    pass
+
+
+AUDIO_DATASET_TRANSFORM_REGISTRY = {}
+AUDIO_DATASET_TRANSFORM_CLASS_NAMES = set()
+
+
+def get_audio_dataset_transform(name):
+    return AUDIO_DATASET_TRANSFORM_REGISTRY[name]
+
+
+def register_audio_dataset_transform(name):
+    return register_audio_transform(
+        name,
+        AudioDatasetTransform,
+        AUDIO_DATASET_TRANSFORM_REGISTRY,
+        AUDIO_DATASET_TRANSFORM_CLASS_NAMES,
+    )
+
+
+import_transforms(os.path.dirname(__file__), "dataset")
+
+
+class CompositeAudioDatasetTransform(CompositeAudioTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        return super()._from_config_dict(
+            cls,
+            "dataset",
+            get_audio_dataset_transform,
+            CompositeAudioDatasetTransform,
+            config,
+            return_empty=True,
+        )
+
+    def get_transform(self, cls):
+        for t in self.transforms:
+            if isinstance(t, cls):
+                return t
+        return None
+
+    def has_transform(self, cls):
+        return self.get_transform(cls) is not None

modules/voice_conversion/fairseq/data/audio/dataset_transforms/concataugment.py

@@ -0,0 +1,61 @@
+from typing import List
+import numpy as np
+
+from fairseq.data.audio.dataset_transforms import (
+    AudioDatasetTransform,
+    register_audio_dataset_transform,
+)
+
+_DEFAULTS = {"rate": 0.25, "max_tokens": 3000, "attempts": 5}
+
+
+@register_audio_dataset_transform("concataugment")
+class ConcatAugment(AudioDatasetTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return ConcatAugment(
+            _config.get("rate", _DEFAULTS["rate"]),
+            _config.get("max_tokens", _DEFAULTS["max_tokens"]),
+            _config.get("attempts", _DEFAULTS["attempts"]),
+        )
+
+    def __init__(
+        self,
+        rate=_DEFAULTS["rate"],
+        max_tokens=_DEFAULTS["max_tokens"],
+        attempts=_DEFAULTS["attempts"],
+    ):
+        self.rate, self.max_tokens, self.attempts = rate, max_tokens, attempts
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + "("
+            + ", ".join(
+                [
+                    f"rate={self.rate}",
+                    f"max_tokens={self.max_tokens}",
+                    f"attempts={self.attempts}",
+                ]
+            )
+            + ")"
+        )
+
+    def find_indices(self, index: int, n_frames: List[int], n_samples: int):
+        # skip conditions: application rate, max_tokens limit exceeded
+        if np.random.random() > self.rate:
+            return [index]
+        if self.max_tokens and n_frames[index] > self.max_tokens:
+            return [index]
+
+        # pick second sample to concatenate
+        for _ in range(self.attempts):
+            index2 = np.random.randint(0, n_samples)
+            if index2 != index and (
+                not self.max_tokens
+                or n_frames[index] + n_frames[index2] < self.max_tokens
+            ):
+                return [index, index2]
+
+        return [index]

modules/voice_conversion/fairseq/data/audio/dataset_transforms/noisyoverlapaugment.py

@@ -0,0 +1,105 @@
+import numpy as np
+import torch
+
+from fairseq.data.audio import rand_uniform
+from fairseq.data.audio.dataset_transforms import (
+    AudioDatasetTransform,
+    register_audio_dataset_transform,
+)
+from fairseq.data.audio.waveform_transforms.noiseaugment import (
+    NoiseAugmentTransform,
+)
+
+_DEFAULTS = {
+    "rate": 0.25,
+    "mixing_noise_rate": 0.1,
+    "noise_path": "",
+    "noise_snr_min": -5,
+    "noise_snr_max": 5,
+    "utterance_snr_min": -5,
+    "utterance_snr_max": 5,
+}
+
+
+@register_audio_dataset_transform("noisyoverlapaugment")
+class NoisyOverlapAugment(AudioDatasetTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return NoisyOverlapAugment(
+            _config.get("rate", _DEFAULTS["rate"]),
+            _config.get("mixing_noise_rate", _DEFAULTS["mixing_noise_rate"]),
+            _config.get("noise_path", _DEFAULTS["noise_path"]),
+            _config.get("noise_snr_min", _DEFAULTS["noise_snr_min"]),
+            _config.get("noise_snr_max", _DEFAULTS["noise_snr_max"]),
+            _config.get("utterance_snr_min", _DEFAULTS["utterance_snr_min"]),
+            _config.get("utterance_snr_max", _DEFAULTS["utterance_snr_max"]),
+        )
+
+    def __init__(
+        self,
+        rate=_DEFAULTS["rate"],
+        mixing_noise_rate=_DEFAULTS["mixing_noise_rate"],
+        noise_path=_DEFAULTS["noise_path"],
+        noise_snr_min=_DEFAULTS["noise_snr_min"],
+        noise_snr_max=_DEFAULTS["noise_snr_max"],
+        utterance_snr_min=_DEFAULTS["utterance_snr_min"],
+        utterance_snr_max=_DEFAULTS["utterance_snr_max"],
+    ):
+        self.rate = rate
+        self.mixing_noise_rate = mixing_noise_rate
+        self.noise_shaper = NoiseAugmentTransform(noise_path)
+        self.noise_snr_min = noise_snr_min
+        self.noise_snr_max = noise_snr_max
+        self.utterance_snr_min = utterance_snr_min
+        self.utterance_snr_max = utterance_snr_max
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + "("
+            + ", ".join(
+                [
+                    f"rate={self.rate}",
+                    f"mixing_noise_rate={self.mixing_noise_rate}",
+                    f"noise_snr_min={self.noise_snr_min}",
+                    f"noise_snr_max={self.noise_snr_max}",
+                    f"utterance_snr_min={self.utterance_snr_min}",
+                    f"utterance_snr_max={self.utterance_snr_max}",
+                ]
+            )
+            + ")"
+        )
+
+    def __call__(self, sources):
+        for i, source in enumerate(sources):
+            if np.random.random() > self.rate:
+                continue
+
+            pri = source.numpy()
+
+            if np.random.random() > self.mixing_noise_rate:
+                sec = sources[np.random.randint(0, len(sources))].numpy()
+                snr = rand_uniform(self.utterance_snr_min, self.utterance_snr_max)
+            else:
+                sec = self.noise_shaper.pick_sample(source.shape)
+                snr = rand_uniform(self.noise_snr_min, self.noise_snr_max)
+
+            L1 = pri.shape[-1]
+            L2 = sec.shape[-1]
+            l = np.random.randint(0, min(round(L1 / 2), L2))  # mix len
+            s_source = np.random.randint(0, L1 - l)
+            s_sec = np.random.randint(0, L2 - l)
+
+            get_power = lambda x: np.mean(x**2)
+            if get_power(sec) == 0:
+                continue
+
+            scl = np.sqrt(get_power(pri) / (np.power(10, snr / 10) * get_power(sec)))
+
+            pri[s_source : s_source + l] = np.add(
+                pri[s_source : s_source + l], np.multiply(scl, sec[s_sec : s_sec + l])
+            )
+            sources[i] = torch.from_numpy(pri).float()
+
+        return sources

modules/voice_conversion/fairseq/data/audio/feature_transforms/__init__.py

@@ -0,0 +1,43 @@
+import os
+from fairseq.data.audio import (
+    AudioTransform,
+    CompositeAudioTransform,
+    import_transforms,
+    register_audio_transform,
+)
+
+
+class AudioFeatureTransform(AudioTransform):
+    pass
+
+
+AUDIO_FEATURE_TRANSFORM_REGISTRY = {}
+AUDIO_FEATURE_TRANSFORM_CLASS_NAMES = set()
+
+
+def get_audio_feature_transform(name):
+    return AUDIO_FEATURE_TRANSFORM_REGISTRY[name]
+
+
+def register_audio_feature_transform(name):
+    return register_audio_transform(
+        name,
+        AudioFeatureTransform,
+        AUDIO_FEATURE_TRANSFORM_REGISTRY,
+        AUDIO_FEATURE_TRANSFORM_CLASS_NAMES,
+    )
+
+
+import_transforms(os.path.dirname(__file__), "feature")
+
+
+class CompositeAudioFeatureTransform(CompositeAudioTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        return super()._from_config_dict(
+            cls,
+            "feature",
+            get_audio_feature_transform,
+            CompositeAudioFeatureTransform,
+            config,
+        )

modules/voice_conversion/fairseq/data/audio/feature_transforms/delta_deltas.py

@@ -0,0 +1,37 @@
+import numpy as np
+import torch
+from fairseq.data.audio.feature_transforms import (
+    AudioFeatureTransform,
+    register_audio_feature_transform,
+)
+
+
+@register_audio_feature_transform("delta_deltas")
+class DeltaDeltas(AudioFeatureTransform):
+    """Expand delta-deltas features from spectrum."""
+
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return DeltaDeltas(_config.get("win_length", 5))
+
+    def __init__(self, win_length=5):
+        self.win_length = win_length
+
+    def __repr__(self):
+        return self.__class__.__name__
+
+    def __call__(self, spectrogram):
+        from torchaudio.functional import compute_deltas
+
+        assert len(spectrogram.shape) == 2, "spectrogram must be a 2-D tensor."
+        # spectrogram is T x F, while compute_deltas takes (…, F, T)
+        spectrogram = torch.from_numpy(spectrogram).transpose(0, 1)
+        delta = compute_deltas(spectrogram)
+        delta_delta = compute_deltas(delta)
+
+        out_feat = np.concatenate(
+            [spectrogram, delta.numpy(), delta_delta.numpy()], axis=0
+        )
+        out_feat = np.transpose(out_feat)
+        return out_feat

modules/voice_conversion/fairseq/data/audio/feature_transforms/global_cmvn.py

@@ -0,0 +1,29 @@
+import numpy as np
+from fairseq.data.audio.feature_transforms import (
+    AudioFeatureTransform,
+    register_audio_feature_transform,
+)
+
+
+@register_audio_feature_transform("global_cmvn")
+class GlobalCMVN(AudioFeatureTransform):
+    """Global CMVN (cepstral mean and variance normalization). The global mean
+    and variance need to be pre-computed and stored in NumPy format (.npz)."""
+
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return GlobalCMVN(_config.get("stats_npz_path"))
+
+    def __init__(self, stats_npz_path):
+        self.stats_npz_path = stats_npz_path
+        stats = np.load(stats_npz_path)
+        self.mean, self.std = stats["mean"], stats["std"]
+
+    def __repr__(self):
+        return self.__class__.__name__ + f'(stats_npz_path="{self.stats_npz_path}")'
+
+    def __call__(self, x):
+        x = np.subtract(x, self.mean)
+        x = np.divide(x, self.std)
+        return x

modules/voice_conversion/fairseq/data/audio/feature_transforms/specaugment.py

@@ -0,0 +1,131 @@
+import math
+import numbers
+from typing import Optional
+
+import numpy as np
+from fairseq.data.audio.feature_transforms import (
+    AudioFeatureTransform,
+    register_audio_feature_transform,
+)
+
+
+@register_audio_feature_transform("specaugment")
+class SpecAugmentTransform(AudioFeatureTransform):
+    """SpecAugment (https://arxiv.org/abs/1904.08779)"""
+
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return SpecAugmentTransform(
+            _config.get("time_warp_W", 0),
+            _config.get("freq_mask_N", 0),
+            _config.get("freq_mask_F", 0),
+            _config.get("time_mask_N", 0),
+            _config.get("time_mask_T", 0),
+            _config.get("time_mask_p", 0.0),
+            _config.get("mask_value", None),
+        )
+
+    def __init__(
+        self,
+        time_warp_w: int = 0,
+        freq_mask_n: int = 0,
+        freq_mask_f: int = 0,
+        time_mask_n: int = 0,
+        time_mask_t: int = 0,
+        time_mask_p: float = 0.0,
+        mask_value: Optional[float] = 0.0,
+    ):
+        # Sanity checks
+        assert mask_value is None or isinstance(
+            mask_value, numbers.Number
+        ), f"mask_value (type: {type(mask_value)}) must be None or a number"
+        if freq_mask_n > 0:
+            assert freq_mask_f > 0, (
+                f"freq_mask_F ({freq_mask_f}) "
+                f"must be larger than 0 when doing freq masking."
+            )
+        if time_mask_n > 0:
+            assert time_mask_t > 0, (
+                f"time_mask_T ({time_mask_t}) must be larger than 0 when "
+                f"doing time masking."
+            )
+
+        self.time_warp_w = time_warp_w
+        self.freq_mask_n = freq_mask_n
+        self.freq_mask_f = freq_mask_f
+        self.time_mask_n = time_mask_n
+        self.time_mask_t = time_mask_t
+        self.time_mask_p = time_mask_p
+        self.mask_value = mask_value
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + "("
+            + ", ".join(
+                [
+                    f"time_warp_w={self.time_warp_w}",
+                    f"freq_mask_n={self.freq_mask_n}",
+                    f"freq_mask_f={self.freq_mask_f}",
+                    f"time_mask_n={self.time_mask_n}",
+                    f"time_mask_t={self.time_mask_t}",
+                    f"time_mask_p={self.time_mask_p}",
+                ]
+            )
+            + ")"
+        )
+
+    def __call__(self, spectrogram):
+        assert len(spectrogram.shape) == 2, "spectrogram must be a 2-D tensor."
+
+        distorted = spectrogram.copy()  # make a copy of input spectrogram.
+        num_frames = spectrogram.shape[0]  # or 'tau' in the paper.
+        num_freqs = spectrogram.shape[1]  # or 'miu' in the paper.
+        mask_value = self.mask_value
+
+        if mask_value is None:  # if no value was specified, use local mean.
+            mask_value = spectrogram.mean()
+
+        if num_frames == 0:
+            return spectrogram
+
+        if num_freqs < self.freq_mask_f:
+            return spectrogram
+
+        if self.time_warp_w > 0:
+            if 2 * self.time_warp_w < num_frames:
+                import cv2
+
+                w0 = np.random.randint(self.time_warp_w, num_frames - self.time_warp_w)
+                w = np.random.randint(-self.time_warp_w + 1, self.time_warp_w)
+                upper, lower = distorted[:w0, :], distorted[w0:, :]
+                upper = cv2.resize(
+                    upper, dsize=(num_freqs, w0 + w), interpolation=cv2.INTER_LINEAR
+                )
+                lower = cv2.resize(
+                    lower,
+                    dsize=(num_freqs, num_frames - w0 - w),
+                    interpolation=cv2.INTER_LINEAR,
+                )
+                distorted = np.concatenate((upper, lower), axis=0)
+
+        for _i in range(self.freq_mask_n):
+            f = np.random.randint(0, self.freq_mask_f)
+            f0 = np.random.randint(0, num_freqs - f)
+            if f != 0:
+                distorted[:, f0 : f0 + f] = mask_value
+
+        max_time_mask_t = min(
+            self.time_mask_t, math.floor(num_frames * self.time_mask_p)
+        )
+        if max_time_mask_t < 1:
+            return distorted
+
+        for _i in range(self.time_mask_n):
+            t = np.random.randint(0, max_time_mask_t)
+            t0 = np.random.randint(0, num_frames - t)
+            if t != 0:
+                distorted[t0 : t0 + t, :] = mask_value
+
+        return distorted

modules/voice_conversion/fairseq/data/audio/feature_transforms/utterance_cmvn.py

@@ -0,0 +1,41 @@
+import numpy as np
+
+from fairseq.data.audio.feature_transforms import (
+    AudioFeatureTransform,
+    register_audio_feature_transform,
+)
+
+
+@register_audio_feature_transform("utterance_cmvn")
+class UtteranceCMVN(AudioFeatureTransform):
+    """Utterance-level CMVN (cepstral mean and variance normalization)"""
+
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return UtteranceCMVN(
+            _config.get("norm_means", True),
+            _config.get("norm_vars", True),
+        )
+
+    def __init__(self, norm_means=True, norm_vars=True):
+        self.norm_means, self.norm_vars = norm_means, norm_vars
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + f"(norm_means={self.norm_means}, norm_vars={self.norm_vars})"
+        )
+
+    def __call__(self, x):
+        mean = x.mean(axis=0)
+        square_sums = (x**2).sum(axis=0)
+
+        if self.norm_means:
+            x = np.subtract(x, mean)
+        if self.norm_vars:
+            var = square_sums / x.shape[0] - mean**2
+            std = np.sqrt(np.maximum(var, 1e-10))
+            x = np.divide(x, std)
+
+        return x

modules/voice_conversion/fairseq/data/audio/frm_text_to_speech_dataset.py

@@ -0,0 +1,205 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.abs
+
+import csv
+import logging
+import os.path as op
+from typing import List, Optional
+
+import numpy as np
+import torch
+from fairseq.data import Dictionary
+from fairseq.data.audio.speech_to_text_dataset import S2TDataConfig
+from fairseq.data.audio.text_to_speech_dataset import (
+    TextToSpeechDataset,
+    TextToSpeechDatasetCreator,
+)
+
+logger = logging.getLogger(__name__)
+
+
+class FrmTextToSpeechDataset(TextToSpeechDataset):
+    def __init__(
+        self,
+        split: str,
+        is_train_split: bool,
+        data_cfg: S2TDataConfig,
+        audio_paths: List[str],
+        n_frames: List[int],
+        src_texts: Optional[List[str]] = None,
+        tgt_texts: Optional[List[str]] = None,
+        speakers: Optional[List[str]] = None,
+        src_langs: Optional[List[str]] = None,
+        tgt_langs: Optional[List[str]] = None,
+        ids: Optional[List[str]] = None,
+        tgt_dict: Optional[Dictionary] = None,
+        pre_tokenizer=None,
+        bpe_tokenizer=None,
+        n_frames_per_step=1,
+        speaker_to_id=None,
+        do_chunk=False,
+        chunk_bound=-1,
+        chunk_init=50,
+        chunk_incr=5,
+        add_eos=True,
+        dedup=True,
+        ref_fpu=-1,
+    ):
+        # It assumes texts are encoded at a fixed frame-rate
+        super().__init__(
+            split=split,
+            is_train_split=is_train_split,
+            data_cfg=data_cfg,
+            audio_paths=audio_paths,
+            n_frames=n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            n_frames_per_step=n_frames_per_step,
+            speaker_to_id=speaker_to_id,
+        )
+
+        self.do_chunk = do_chunk
+        self.chunk_bound = chunk_bound
+        self.chunk_init = chunk_init
+        self.chunk_incr = chunk_incr
+        self.add_eos = add_eos
+        self.dedup = dedup
+        self.ref_fpu = ref_fpu
+
+        self.chunk_size = -1
+
+        if do_chunk:
+            assert self.chunk_incr >= 0
+            assert self.pre_tokenizer is None
+
+    def __getitem__(self, index):
+        index, source, target, speaker_id, _, _, _ = super().__getitem__(index)
+        if target[-1].item() == self.tgt_dict.eos_index:
+            target = target[:-1]
+
+        fpu = source.size(0) / target.size(0)  # frame-per-unit
+        fps = self.n_frames_per_step
+        assert (
+            self.ref_fpu == -1 or abs((fpu * fps - self.ref_fpu) / self.ref_fpu) < 0.1
+        ), f"{fpu*fps} != {self.ref_fpu}"
+
+        # only chunk training split
+        if self.is_train_split and self.do_chunk and self.chunk_size > 0:
+            lang = target[: int(self.data_cfg.prepend_tgt_lang_tag)]
+            text = target[int(self.data_cfg.prepend_tgt_lang_tag) :]
+            size = len(text)
+            chunk_size = min(self.chunk_size, size)
+            chunk_start = np.random.randint(size - chunk_size + 1)
+            text = text[chunk_start : chunk_start + chunk_size]
+            target = torch.cat((lang, text), 0)
+
+            f_size = int(np.floor(chunk_size * fpu))
+            f_start = int(np.floor(chunk_start * fpu))
+            assert f_size > 0
+            source = source[f_start : f_start + f_size, :]
+
+        if self.dedup:
+            target = torch.unique_consecutive(target)
+
+        if self.add_eos:
+            eos_idx = self.tgt_dict.eos_index
+            target = torch.cat((target, torch.LongTensor([eos_idx])), 0)
+
+        return index, source, target, speaker_id
+
+    def set_epoch(self, epoch):
+        if self.is_train_split and self.do_chunk:
+            old = self.chunk_size
+            self.chunk_size = self.chunk_init + epoch * self.chunk_incr
+            if self.chunk_bound > 0:
+                self.chunk_size = min(self.chunk_size, self.chunk_bound)
+            logger.info(
+                (
+                    f"{self.split}: setting chunk size "
+                    f"from {old} to {self.chunk_size}"
+                )
+            )
+
+
+class FrmTextToSpeechDatasetCreator(TextToSpeechDatasetCreator):
+    # inherit for key names
+    @classmethod
+    def from_tsv(
+        cls,
+        root: str,
+        data_cfg: S2TDataConfig,
+        split: str,
+        tgt_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        is_train_split: bool,
+        n_frames_per_step: int,
+        speaker_to_id,
+        do_chunk: bool = False,
+        chunk_bound: int = -1,
+        chunk_init: int = 50,
+        chunk_incr: int = 5,
+        add_eos: bool = True,
+        dedup: bool = True,
+        ref_fpu: float = -1,
+    ) -> FrmTextToSpeechDataset:
+        tsv_path = op.join(root, f"{split}.tsv")
+        if not op.isfile(tsv_path):
+            raise FileNotFoundError(f"Dataset not found: {tsv_path}")
+        with open(tsv_path) as f:
+            reader = csv.DictReader(
+                f,
+                delimiter="\t",
+                quotechar=None,
+                doublequote=False,
+                lineterminator="\n",
+                quoting=csv.QUOTE_NONE,
+            )
+            s = [dict(e) for e in reader]
+            assert len(s) > 0
+
+        ids = [ss[cls.KEY_ID] for ss in s]
+        audio_paths = [op.join(data_cfg.audio_root, ss[cls.KEY_AUDIO]) for ss in s]
+        n_frames = [int(ss[cls.KEY_N_FRAMES]) for ss in s]
+        tgt_texts = [ss[cls.KEY_TGT_TEXT] for ss in s]
+        src_texts = [ss.get(cls.KEY_SRC_TEXT, cls.DEFAULT_SRC_TEXT) for ss in s]
+        speakers = [ss.get(cls.KEY_SPEAKER, cls.DEFAULT_SPEAKER) for ss in s]
+        src_langs = [ss.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for ss in s]
+        tgt_langs = [ss.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for ss in s]
+
+        return FrmTextToSpeechDataset(
+            split=split,
+            is_train_split=is_train_split,
+            data_cfg=data_cfg,
+            audio_paths=audio_paths,
+            n_frames=n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            n_frames_per_step=n_frames_per_step,
+            speaker_to_id=speaker_to_id,
+            do_chunk=do_chunk,
+            chunk_bound=chunk_bound,
+            chunk_init=chunk_init,
+            chunk_incr=chunk_incr,
+            add_eos=add_eos,
+            dedup=dedup,
+            ref_fpu=ref_fpu,
+        )

modules/voice_conversion/fairseq/data/audio/hubert_dataset.py

@@ -0,0 +1,356 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+import itertools
+import logging
+import os
+import sys
+from typing import Any, List, Optional, Union
+
+import numpy as np
+
+import torch
+import torch.nn.functional as F
+from fairseq.data import data_utils
+from fairseq.data.fairseq_dataset import FairseqDataset
+from fairseq.data.audio.audio_utils import (
+    parse_path,
+    read_from_stored_zip,
+)
+import io
+
+logger = logging.getLogger(__name__)
+
+
+def load_audio(manifest_path, max_keep, min_keep):
+    n_long, n_short = 0, 0
+    names, inds, sizes = [], [], []
+    with open(manifest_path) as f:
+        root = f.readline().strip()
+        for ind, line in enumerate(f):
+            items = line.strip().split("\t")
+            assert len(items) == 2, line
+            sz = int(items[1])
+            if min_keep is not None and sz < min_keep:
+                n_short += 1
+            elif max_keep is not None and sz > max_keep:
+                n_long += 1
+            else:
+                names.append(items[0])
+                inds.append(ind)
+                sizes.append(sz)
+    tot = ind + 1
+    logger.info(
+        (
+            f"max_keep={max_keep}, min_keep={min_keep}, "
+            f"loaded {len(names)}, skipped {n_short} short and {n_long} long, "
+            f"longest-loaded={max(sizes)}, shortest-loaded={min(sizes)}"
+        )
+    )
+    return root, names, inds, tot, sizes
+
+
+def load_label(label_path, inds, tot):
+    with open(label_path) as f:
+        labels = [line.rstrip() for line in f]
+        assert (
+            len(labels) == tot
+        ), f"number of labels does not match ({len(labels)} != {tot})"
+        labels = [labels[i] for i in inds]
+    return labels
+
+
+def load_label_offset(label_path, inds, tot):
+    with open(label_path) as f:
+        code_lengths = [len(line.encode("utf-8")) for line in f]
+        assert (
+            len(code_lengths) == tot
+        ), f"number of labels does not match ({len(code_lengths)} != {tot})"
+        offsets = list(itertools.accumulate([0] + code_lengths))
+        offsets = [(offsets[i], offsets[i + 1]) for i in inds]
+    return offsets
+
+
+def verify_label_lengths(
+    audio_sizes,
+    audio_rate,
+    label_path,
+    label_rate,
+    inds,
+    tot,
+    tol=0.1,  # tolerance in seconds
+):
+    if label_rate < 0:
+        logger.info(f"{label_path} is sequence label. skipped")
+        return
+
+    with open(label_path) as f:
+        lengths = [len(line.rstrip().split()) for line in f]
+        assert len(lengths) == tot
+        lengths = [lengths[i] for i in inds]
+    num_invalid = 0
+    for i, ind in enumerate(inds):
+        dur_from_audio = audio_sizes[i] / audio_rate
+        dur_from_label = lengths[i] / label_rate
+        if abs(dur_from_audio - dur_from_label) > tol:
+            logger.warning(
+                (
+                    f"audio and label duration differ too much "
+                    f"(|{dur_from_audio} - {dur_from_label}| > {tol}) "
+                    f"in line {ind+1} of {label_path}. Check if `label_rate` "
+                    f"is correctly set (currently {label_rate}). "
+                    f"num. of samples = {audio_sizes[i]}; "
+                    f"label length = {lengths[i]}"
+                )
+            )
+            num_invalid += 1
+    if num_invalid > 0:
+        logger.warning(
+            f"total {num_invalid} (audio, label) pairs with mismatched lengths"
+        )
+
+
+class HubertDataset(FairseqDataset):
+    def __init__(
+        self,
+        manifest_path: str,
+        sample_rate: float,
+        label_paths: List[str],
+        label_rates: Union[List[float], float],  # -1 for sequence labels
+        pad_list: List[str],
+        eos_list: List[str],
+        label_processors: Optional[List[Any]] = None,
+        max_keep_sample_size: Optional[int] = None,
+        min_keep_sample_size: Optional[int] = None,
+        max_sample_size: Optional[int] = None,
+        shuffle: bool = True,
+        pad_audio: bool = False,
+        normalize: bool = False,
+        store_labels: bool = True,
+        random_crop: bool = False,
+        single_target: bool = False,
+    ):
+        self.audio_root, self.audio_names, inds, tot, self.sizes = load_audio(
+            manifest_path, max_keep_sample_size, min_keep_sample_size
+        )
+        self.sample_rate = sample_rate
+        self.shuffle = shuffle
+        self.random_crop = random_crop
+
+        self.num_labels = len(label_paths)
+        self.pad_list = pad_list
+        self.eos_list = eos_list
+        self.label_processors = label_processors
+        self.single_target = single_target
+        self.label_rates = (
+            [label_rates for _ in range(len(label_paths))]
+            if isinstance(label_rates, float)
+            else label_rates
+        )
+        self.store_labels = store_labels
+        if store_labels:
+            self.label_list = [load_label(p, inds, tot) for p in label_paths]
+        else:
+            self.label_paths = label_paths
+            self.label_offsets_list = [
+                load_label_offset(p, inds, tot) for p in label_paths
+            ]
+        assert label_processors is None or len(label_processors) == self.num_labels
+        for label_path, label_rate in zip(label_paths, self.label_rates):
+            verify_label_lengths(
+                self.sizes, sample_rate, label_path, label_rate, inds, tot
+            )
+
+        self.max_sample_size = (
+            max_sample_size if max_sample_size is not None else sys.maxsize
+        )
+        self.pad_audio = pad_audio
+        self.normalize = normalize
+        logger.info(
+            f"pad_audio={pad_audio}, random_crop={random_crop}, "
+            f"normalize={normalize}, max_sample_size={self.max_sample_size}"
+        )
+
+    def get_audio(self, index):
+        import soundfile as sf
+
+        wav_path = os.path.join(self.audio_root, self.audio_names[index])
+        _path, slice_ptr = parse_path(wav_path)
+        if len(slice_ptr) == 0:
+            wav, cur_sample_rate = sf.read(_path)
+        else:
+            assert _path.endswith(".zip")
+            data = read_from_stored_zip(_path, slice_ptr[0], slice_ptr[1])
+            f = io.BytesIO(data)
+            wav, cur_sample_rate = sf.read(f)
+        wav = torch.from_numpy(wav).float()
+        wav = self.postprocess(wav, cur_sample_rate)
+        return wav
+
+    def get_label(self, index, label_idx):
+        if self.store_labels:
+            label = self.label_list[label_idx][index]
+        else:
+            with open(self.label_paths[label_idx]) as f:
+                offset_s, offset_e = self.label_offsets_list[label_idx][index]
+                f.seek(offset_s)
+                label = f.read(offset_e - offset_s)
+
+        if self.label_processors is not None:
+            label = self.label_processors[label_idx](label)
+        return label
+
+    def get_labels(self, index):
+        return [self.get_label(index, i) for i in range(self.num_labels)]
+
+    def __getitem__(self, index):
+        wav = self.get_audio(index)
+        labels = self.get_labels(index)
+        return {"id": index, "source": wav, "label_list": labels}
+
+    def __len__(self):
+        return len(self.sizes)
+
+    def crop_to_max_size(self, wav, target_size):
+        size = len(wav)
+        diff = size - target_size
+        if diff <= 0:
+            return wav, 0
+
+        start, end = 0, target_size
+        if self.random_crop:
+            start = np.random.randint(0, diff + 1)
+            end = size - diff + start
+        return wav[start:end], start
+
+    def collater(self, samples):
+        # target = max(sizes) -> random_crop not used
+        # target = max_sample_size -> random_crop used for long
+        samples = [s for s in samples if s["source"] is not None]
+        if len(samples) == 0:
+            return {}
+
+        audios = [s["source"] for s in samples]
+        audio_sizes = [len(s) for s in audios]
+        if self.pad_audio:
+            audio_size = min(max(audio_sizes), self.max_sample_size)
+        else:
+            audio_size = min(min(audio_sizes), self.max_sample_size)
+        collated_audios, padding_mask, audio_starts = self.collater_audio(
+            audios, audio_size
+        )
+
+        targets_by_label = [
+            [s["label_list"][i] for s in samples] for i in range(self.num_labels)
+        ]
+        targets_list, lengths_list, ntokens_list = self.collater_label(
+            targets_by_label, audio_size, audio_starts
+        )
+
+        net_input = {"source": collated_audios, "padding_mask": padding_mask}
+        batch = {
+            "id": torch.LongTensor([s["id"] for s in samples]),
+            "net_input": net_input,
+        }
+
+        if self.single_target:
+            batch["target_lengths"] = lengths_list[0]
+            batch["ntokens"] = ntokens_list[0]
+            batch["target"] = targets_list[0]
+        else:
+            batch["target_lengths_list"] = lengths_list
+            batch["ntokens_list"] = ntokens_list
+            batch["target_list"] = targets_list
+        return batch
+
+    def collater_audio(self, audios, audio_size):
+        collated_audios = audios[0].new_zeros(len(audios), audio_size)
+        padding_mask = (
+            torch.BoolTensor(collated_audios.shape).fill_(False)
+            # if self.pad_audio else None
+        )
+        audio_starts = [0 for _ in audios]
+        for i, audio in enumerate(audios):
+            diff = len(audio) - audio_size
+            if diff == 0:
+                collated_audios[i] = audio
+            elif diff < 0:
+                assert self.pad_audio
+                collated_audios[i] = torch.cat([audio, audio.new_full((-diff,), 0.0)])
+                padding_mask[i, diff:] = True
+            else:
+                collated_audios[i], audio_starts[i] = self.crop_to_max_size(
+                    audio, audio_size
+                )
+        return collated_audios, padding_mask, audio_starts
+
+    def collater_frm_label(self, targets, audio_size, audio_starts, label_rate, pad):
+        assert label_rate > 0
+        s2f = label_rate / self.sample_rate
+        frm_starts = [int(round(s * s2f)) for s in audio_starts]
+        frm_size = int(round(audio_size * s2f))
+        if not self.pad_audio:
+            rem_size = [len(t) - s for t, s in zip(targets, frm_starts)]
+            frm_size = min(frm_size, *rem_size)
+        targets = [t[s : s + frm_size] for t, s in zip(targets, frm_starts)]
+        logger.debug(f"audio_starts={audio_starts}")
+        logger.debug(f"frame_starts={frm_starts}")
+        logger.debug(f"frame_size={frm_size}")
+
+        lengths = torch.LongTensor([len(t) for t in targets])
+        ntokens = lengths.sum().item()
+        targets = data_utils.collate_tokens(targets, pad_idx=pad, left_pad=False)
+        return targets, lengths, ntokens
+
+    def collater_seq_label(self, targets, pad):
+        lengths = torch.LongTensor([len(t) for t in targets])
+        ntokens = lengths.sum().item()
+        targets = data_utils.collate_tokens(targets, pad_idx=pad, left_pad=False)
+        return targets, lengths, ntokens
+
+    def collater_label(self, targets_by_label, audio_size, audio_starts):
+        targets_list, lengths_list, ntokens_list = [], [], []
+        itr = zip(targets_by_label, self.label_rates, self.pad_list)
+        for targets, label_rate, pad in itr:
+            if label_rate == -1.0:
+                targets, lengths, ntokens = self.collater_seq_label(targets, pad)
+            else:
+                targets, lengths, ntokens = self.collater_frm_label(
+                    targets, audio_size, audio_starts, label_rate, pad
+                )
+            targets_list.append(targets)
+            lengths_list.append(lengths)
+            ntokens_list.append(ntokens)
+        return targets_list, lengths_list, ntokens_list
+
+    def num_tokens(self, index):
+        return self.size(index)
+
+    def size(self, index):
+        if self.pad_audio:
+            return self.sizes[index]
+        return min(self.sizes[index], self.max_sample_size)
+
+    def ordered_indices(self):
+        if self.shuffle:
+            order = [np.random.permutation(len(self))]
+        else:
+            order = [np.arange(len(self))]
+
+        order.append(self.sizes)
+        return np.lexsort(order)[::-1]
+
+    def postprocess(self, wav, cur_sample_rate):
+        if wav.dim() == 2:
+            wav = wav.mean(-1)
+        assert wav.dim() == 1, wav.dim()
+
+        if cur_sample_rate != self.sample_rate:
+            raise Exception(f"sr {cur_sample_rate} != {self.sample_rate}")
+
+        if self.normalize:
+            with torch.no_grad():
+                wav = F.layer_norm(wav, wav.shape)
+        return wav

modules/voice_conversion/fairseq/data/audio/multi_modality_dataset.py

@@ -0,0 +1,284 @@
+# Copyright (c) 2021-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.
+
+import logging
+import math
+from typing import List, Optional, NamedTuple
+
+import numpy as np
+from fairseq.data.resampling_dataset import ResamplingDataset
+import torch
+from fairseq.data import (
+    ConcatDataset,
+    LanguagePairDataset,
+    FileAudioDataset,
+    data_utils,
+)
+from fairseq.data import FairseqDataset
+
+logger = logging.getLogger(__name__)
+
+
+class ModalityDatasetItem(NamedTuple):
+    datasetname: str
+    dataset: any
+    max_positions: List[int]
+    max_tokens: Optional[int] = None
+    max_sentences: Optional[int] = None
+
+
+def resampling_dataset_present(ds):
+    if isinstance(ds, ResamplingDataset):
+        return True
+    if isinstance(ds, ConcatDataset):
+        return any(resampling_dataset_present(d) for d in ds.datasets)
+    if hasattr(ds, "dataset"):
+        return resampling_dataset_present(ds.dataset)
+    return False
+
+
+# MultiModalityDataset: it concate multiple datasets with different modalities.
+# Compared with ConcatDataset it can 1) sample data given the ratios for different datasets
+# 2) it adds mode to indicate what type of the data samples come from.
+# It will be used with GroupedEpochBatchIterator together to generate mini-batch with samples
+# from the same type of dataset
+# If only one dataset is used, it will perform like the original dataset with mode added
+class MultiModalityDataset(ConcatDataset):
+    def __init__(self, datasets: List[ModalityDatasetItem]):
+        id_to_mode = []
+        dsets = []
+        max_tokens = []
+        max_sentences = []
+        max_positions = []
+        for dset in datasets:
+            id_to_mode.append(dset.datasetname)
+            dsets.append(dset.dataset)
+            max_tokens.append(dset.max_tokens)
+            max_positions.append(dset.max_positions)
+            max_sentences.append(dset.max_sentences)
+        weights = [1.0 for s in dsets]
+        super().__init__(dsets, weights)
+        self.max_tokens = max_tokens
+        self.max_positions = max_positions
+        self.max_sentences = max_sentences
+        self.id_to_mode = id_to_mode
+        self.raw_sub_batch_samplers = []
+        self._cur_epoch = 0
+
+    def set_epoch(self, epoch):
+        super().set_epoch(epoch)
+        self._cur_epoch = epoch
+
+    def __getitem__(self, idx):
+        dataset_idx, sample_idx = self._get_dataset_and_sample_index(idx)
+        sample = self.datasets[dataset_idx][sample_idx]
+        return (dataset_idx, sample)
+
+    def collater(self, samples):
+        if len(samples) == 0:
+            return {}
+        dataset_idx = samples[0][0]
+        # make sure all samples in samples are from same dataset
+        assert sum([0 if dataset_idx == s[0] else 1 for s in samples]) == 0
+        samples = self.datasets[dataset_idx].collater([x[1] for x in samples])
+        # add mode
+        samples["net_input"]["mode"] = self.id_to_mode[dataset_idx]
+
+        return samples
+
+    def size(self, index: int):
+        if len(self.datasets) == 1:
+            return self.datasets[0].size(index)
+        return super().size(index)
+
+    @property
+    def sizes(self):
+        if len(self.datasets) == 1:
+            return self.datasets[0].sizes
+        return super().sizes
+
+    def ordered_indices(self):
+        """
+        Returns indices sorted by length. So less padding is needed.
+        """
+        if len(self.datasets) == 1:
+            return self.datasets[0].ordered_indices()
+        indices_group = []
+        for d_idx, ds in enumerate(self.datasets):
+            sample_num = self.cumulative_sizes[d_idx]
+            if d_idx > 0:
+                sample_num = sample_num - self.cumulative_sizes[d_idx - 1]
+            assert sample_num == len(ds)
+            indices_group.append(ds.ordered_indices())
+        return indices_group
+
+    def get_raw_batch_samplers(self, required_batch_size_multiple, seed):
+        with data_utils.numpy_seed(seed):
+            indices = self.ordered_indices()
+        for i, ds in enumerate(self.datasets):
+            # If we have ResamplingDataset, the same id can correpond to a different
+            # sample in the next epoch, so we need to rebuild this at every epoch
+            if i < len(self.raw_sub_batch_samplers) and not resampling_dataset_present(
+                ds
+            ):
+                logger.info(f"dataset {i} is valid and it is not re-sampled")
+                continue
+            indices[i] = ds.filter_indices_by_size(
+                indices[i],
+                self.max_positions[i],
+            )[0]
+            sub_batch_sampler = ds.batch_by_size(
+                indices[i],
+                max_tokens=self.max_tokens[i],
+                max_sentences=self.max_sentences[i],
+                required_batch_size_multiple=required_batch_size_multiple,
+            )
+            if i < len(self.raw_sub_batch_samplers):
+                self.raw_sub_batch_samplers[i] = sub_batch_sampler
+            else:
+                self.raw_sub_batch_samplers.append(sub_batch_sampler)
+
+    def get_batch_samplers(self, mult_ratios, required_batch_size_multiple, seed):
+        self.get_raw_batch_samplers(required_batch_size_multiple, seed)
+        batch_samplers = []
+        for i, _ in enumerate(self.datasets):
+            if i > 0:
+                sub_batch_sampler = [
+                    [y + self.cumulative_sizes[i - 1] for y in x]
+                    for x in self.raw_sub_batch_samplers[i]
+                ]
+            else:
+                sub_batch_sampler = list(self.raw_sub_batch_samplers[i])
+            smp_r = mult_ratios[i]
+            if smp_r != 1:
+                is_increase = "increased" if smp_r > 1 else "decreased"
+                logger.info(
+                    "number of batch for the dataset {} is {} from {} to {}".format(
+                        self.id_to_mode[i],
+                        is_increase,
+                        len(sub_batch_sampler),
+                        int(len(sub_batch_sampler) * smp_r),
+                    )
+                )
+                mul_samplers = []
+                for _ in range(math.floor(smp_r)):
+                    mul_samplers = mul_samplers + sub_batch_sampler
+                if math.floor(smp_r) != smp_r:
+                    with data_utils.numpy_seed(seed + self._cur_epoch):
+                        np.random.shuffle(sub_batch_sampler)
+                        smp_num = int(
+                            (smp_r - math.floor(smp_r)) * len(sub_batch_sampler)
+                        )
+                    mul_samplers = mul_samplers + sub_batch_sampler[:smp_num]
+                sub_batch_sampler = mul_samplers
+            else:
+                logger.info(
+                    "dataset {} batch number is {} ".format(
+                        self.id_to_mode[i], len(sub_batch_sampler)
+                    )
+                )
+            batch_samplers.append(sub_batch_sampler)
+
+        return batch_samplers
+
+
+class LangPairMaskDataset(FairseqDataset):
+    def __init__(
+        self,
+        dataset: LanguagePairDataset,
+        src_eos: int,
+        src_bos: Optional[int] = None,
+        noise_id: Optional[int] = -1,
+        mask_ratio: Optional[float] = 0,
+        mask_type: Optional[str] = "random",
+    ):
+        self.dataset = dataset
+        self.src_eos = src_eos
+        self.src_bos = src_bos
+        self.noise_id = noise_id
+        self.mask_ratio = mask_ratio
+        self.mask_type = mask_type
+        assert mask_type in ("random", "tail")
+
+    @property
+    def src_sizes(self):
+        return self.dataset.src_sizes
+
+    @property
+    def tgt_sizes(self):
+        return self.dataset.tgt_sizes
+
+    @property
+    def sizes(self):
+        # dataset.sizes can be a dynamically computed sizes:
+        return self.dataset.sizes
+
+    def get_batch_shapes(self):
+        if hasattr(self.dataset, "get_batch_shapes"):
+            return self.dataset.get_batch_shapes()
+        return self.dataset.buckets
+
+    def num_tokens_vec(self, indices):
+        return self.dataset.num_tokens_vec(indices)
+
+    def __len__(self):
+        return len(self.dataset)
+
+    def num_tokens(self, index):
+        return self.dataset.num_tokens(index)
+
+    def size(self, index):
+        return self.dataset.size(index)
+
+    def ordered_indices(self):
+        return self.dataset.ordered_indices()
+
+    @property
+    def supports_prefetch(self):
+        return getattr(self.dataset, "supports_prefetch", False)
+
+    def prefetch(self, indices):
+        return self.dataset.prefetch(indices)
+
+    def mask_src_tokens(self, sample):
+        src_item = sample["source"]
+        mask = None
+        if self.mask_type == "random":
+            mask = torch.rand(len(src_item)).le(self.mask_ratio)
+        else:
+            mask = torch.ones(len(src_item))
+            mask[: int(len(src_item) * (1 - self.mask_ratio))] = 0
+            mask = mask.eq(1)
+        if src_item[0] == self.src_bos:
+            mask[0] = False
+        if src_item[-1] == self.src_eos:
+            mask[-1] = False
+        mask_src_item = src_item.masked_fill(mask, self.noise_id)
+        smp = {"id": sample["id"], "source": mask_src_item, "target": sample["target"]}
+        return smp
+
+    def __getitem__(self, index):
+        sample = self.dataset[index]
+        if self.mask_ratio > 0:
+            sample = self.mask_src_tokens(sample)
+        return sample
+
+    def collater(self, samples, pad_to_length=None):
+        return self.dataset.collater(samples, pad_to_length)
+
+
+class FileAudioDatasetWrapper(FileAudioDataset):
+    def collater(self, samples):
+        samples = super().collater(samples)
+        if len(samples) == 0:
+            return {}
+        samples["net_input"]["src_tokens"] = samples["net_input"]["source"]
+        samples["net_input"]["prev_output_tokens"] = None
+        del samples["net_input"]["source"]
+        samples["net_input"]["src_lengths"] = None
+        samples["net_input"]["alignment"] = None
+        return samples

modules/voice_conversion/fairseq/data/audio/raw_audio_dataset.py

@@ -0,0 +1,393 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+
+import logging
+import os
+import sys
+import io
+
+import numpy as np
+import torch
+import torch.nn.functional as F
+
+from .. import FairseqDataset
+from ..data_utils import compute_mask_indices, get_buckets, get_bucketed_sizes
+from fairseq.data.audio.audio_utils import (
+    parse_path,
+    read_from_stored_zip,
+    is_sf_audio_data,
+)
+from fairseq.data.text_compressor import TextCompressor, TextCompressionLevel
+
+
+logger = logging.getLogger(__name__)
+
+
+class RawAudioDataset(FairseqDataset):
+    def __init__(
+        self,
+        sample_rate,
+        max_sample_size=None,
+        min_sample_size=0,
+        shuffle=True,
+        pad=False,
+        normalize=False,
+        compute_mask_indices=False,
+        **mask_compute_kwargs,
+    ):
+        super().__init__()
+
+        self.sample_rate = sample_rate
+        self.sizes = []
+        self.max_sample_size = (
+            max_sample_size if max_sample_size is not None else sys.maxsize
+        )
+        self.min_sample_size = min_sample_size
+        self.pad = pad
+        self.shuffle = shuffle
+        self.normalize = normalize
+        self.compute_mask_indices = compute_mask_indices
+        if self.compute_mask_indices:
+            self.mask_compute_kwargs = mask_compute_kwargs
+            self._features_size_map = {}
+            self._C = mask_compute_kwargs["encoder_embed_dim"]
+            self._conv_feature_layers = eval(mask_compute_kwargs["conv_feature_layers"])
+
+    def __getitem__(self, index):
+        raise NotImplementedError()
+
+    def __len__(self):
+        return len(self.sizes)
+
+    def postprocess(self, feats, curr_sample_rate):
+        if feats.dim() == 2:
+            feats = feats.mean(-1)
+
+        if curr_sample_rate != self.sample_rate:
+            raise Exception(f"sample rate: {curr_sample_rate}, need {self.sample_rate}")
+
+        assert feats.dim() == 1, feats.dim()
+
+        if self.normalize:
+            with torch.no_grad():
+                feats = F.layer_norm(feats, feats.shape)
+        return feats
+
+    def crop_to_max_size(self, wav, target_size):
+        size = len(wav)
+        diff = size - target_size
+        if diff <= 0:
+            return wav
+
+        start = np.random.randint(0, diff + 1)
+        end = size - diff + start
+        return wav[start:end]
+
+    def _compute_mask_indices(self, dims, padding_mask):
+        B, T, C = dims
+        mask_indices, mask_channel_indices = None, None
+        if self.mask_compute_kwargs["mask_prob"] > 0:
+            mask_indices = compute_mask_indices(
+                (B, T),
+                padding_mask,
+                self.mask_compute_kwargs["mask_prob"],
+                self.mask_compute_kwargs["mask_length"],
+                self.mask_compute_kwargs["mask_selection"],
+                self.mask_compute_kwargs["mask_other"],
+                min_masks=2,
+                no_overlap=self.mask_compute_kwargs["no_mask_overlap"],
+                min_space=self.mask_compute_kwargs["mask_min_space"],
+            )
+            mask_indices = torch.from_numpy(mask_indices)
+        if self.mask_compute_kwargs["mask_channel_prob"] > 0:
+            mask_channel_indices = compute_mask_indices(
+                (B, C),
+                None,
+                self.mask_compute_kwargs["mask_channel_prob"],
+                self.mask_compute_kwargs["mask_channel_length"],
+                self.mask_compute_kwargs["mask_channel_selection"],
+                self.mask_compute_kwargs["mask_channel_other"],
+                no_overlap=self.mask_compute_kwargs["no_mask_channel_overlap"],
+                min_space=self.mask_compute_kwargs["mask_channel_min_space"],
+            )
+            mask_channel_indices = (
+                torch.from_numpy(mask_channel_indices).unsqueeze(1).expand(-1, T, -1)
+            )
+
+        return mask_indices, mask_channel_indices
+
+    @staticmethod
+    def _bucket_tensor(tensor, num_pad, value):
+        return F.pad(tensor, (0, num_pad), value=value)
+
+    def collater(self, samples):
+        samples = [s for s in samples if s["source"] is not None]
+        if len(samples) == 0:
+            return {}
+
+        sources = [s["source"] for s in samples]
+        sizes = [len(s) for s in sources]
+
+        if self.pad:
+            target_size = min(max(sizes), self.max_sample_size)
+        else:
+            target_size = min(min(sizes), self.max_sample_size)
+
+        collated_sources = sources[0].new_zeros(len(sources), target_size)
+        padding_mask = (
+            torch.BoolTensor(collated_sources.shape).fill_(False) if self.pad else None
+        )
+        for i, (source, size) in enumerate(zip(sources, sizes)):
+            diff = size - target_size
+            if diff == 0:
+                collated_sources[i] = source
+            elif diff < 0:
+                assert self.pad
+                collated_sources[i] = torch.cat(
+                    [source, source.new_full((-diff,), 0.0)]
+                )
+                padding_mask[i, diff:] = True
+            else:
+                collated_sources[i] = self.crop_to_max_size(source, target_size)
+
+        input = {"source": collated_sources}
+        out = {"id": torch.LongTensor([s["id"] for s in samples])}
+        if self.pad:
+            input["padding_mask"] = padding_mask
+
+        if hasattr(self, "num_buckets") and self.num_buckets > 0:
+            assert self.pad, "Cannot bucket without padding first."
+            bucket = max(self._bucketed_sizes[s["id"]] for s in samples)
+            num_pad = bucket - collated_sources.size(-1)
+            if num_pad:
+                input["source"] = self._bucket_tensor(collated_sources, num_pad, 0)
+                input["padding_mask"] = self._bucket_tensor(padding_mask, num_pad, True)
+
+        if self.compute_mask_indices:
+            B = input["source"].size(0)
+            T = self._get_mask_indices_dims(input["source"].size(-1))
+            padding_mask_reshaped = input["padding_mask"].clone()
+            extra = padding_mask_reshaped.size(1) % T
+            if extra > 0:
+                padding_mask_reshaped = padding_mask_reshaped[:, :-extra]
+            padding_mask_reshaped = padding_mask_reshaped.view(
+                padding_mask_reshaped.size(0), T, -1
+            )
+            padding_mask_reshaped = padding_mask_reshaped.all(-1)
+            input["padding_count"] = padding_mask_reshaped.sum(-1).max().item()
+            mask_indices, mask_channel_indices = self._compute_mask_indices(
+                (B, T, self._C),
+                padding_mask_reshaped,
+            )
+            input["mask_indices"] = mask_indices
+            input["mask_channel_indices"] = mask_channel_indices
+            out["sample_size"] = mask_indices.sum().item()
+
+        out["net_input"] = input
+        return out
+
+    def _get_mask_indices_dims(self, size, padding=0, dilation=1):
+        if size not in self._features_size_map:
+            L_in = size
+            for (_, kernel_size, stride) in self._conv_feature_layers:
+                L_out = L_in + 2 * padding - dilation * (kernel_size - 1) - 1
+                L_out = 1 + L_out // stride
+                L_in = L_out
+            self._features_size_map[size] = L_out
+        return self._features_size_map[size]
+
+    def num_tokens(self, index):
+        return self.size(index)
+
+    def size(self, index):
+        """Return an example's size as a float or tuple. This value is used when
+        filtering a dataset with ``--max-positions``."""
+        if self.pad:
+            return self.sizes[index]
+        return min(self.sizes[index], self.max_sample_size)
+
+    def ordered_indices(self):
+        """Return an ordered list of indices. Batches will be constructed based
+        on this order."""
+
+        if self.shuffle:
+            order = [np.random.permutation(len(self))]
+            order.append(
+                np.minimum(
+                    np.array(self.sizes),
+                    self.max_sample_size,
+                )
+            )
+            return np.lexsort(order)[::-1]
+        else:
+            return np.arange(len(self))
+
+    def set_bucket_info(self, num_buckets):
+        self.num_buckets = num_buckets
+        if self.num_buckets > 0:
+            self._collated_sizes = np.minimum(
+                np.array(self.sizes),
+                self.max_sample_size,
+            )
+            self.buckets = get_buckets(
+                self._collated_sizes,
+                self.num_buckets,
+            )
+            self._bucketed_sizes = get_bucketed_sizes(
+                self._collated_sizes, self.buckets
+            )
+            logger.info(
+                f"{len(self.buckets)} bucket(s) for the audio dataset: "
+                f"{self.buckets}"
+            )
+
+
+class FileAudioDataset(RawAudioDataset):
+    def __init__(
+        self,
+        manifest_path,
+        sample_rate,
+        max_sample_size=None,
+        min_sample_size=0,
+        shuffle=True,
+        pad=False,
+        normalize=False,
+        num_buckets=0,
+        compute_mask_indices=False,
+        text_compression_level=TextCompressionLevel.none,
+        **mask_compute_kwargs,
+    ):
+        super().__init__(
+            sample_rate=sample_rate,
+            max_sample_size=max_sample_size,
+            min_sample_size=min_sample_size,
+            shuffle=shuffle,
+            pad=pad,
+            normalize=normalize,
+            compute_mask_indices=compute_mask_indices,
+            **mask_compute_kwargs,
+        )
+
+        self.text_compressor = TextCompressor(level=text_compression_level)
+
+        skipped = 0
+        self.fnames = []
+        sizes = []
+        self.skipped_indices = set()
+
+        with open(manifest_path, "r") as f:
+            self.root_dir = f.readline().strip()
+            for i, line in enumerate(f):
+                items = line.strip().split("\t")
+                assert len(items) == 2, line
+                sz = int(items[1])
+                if min_sample_size is not None and sz < min_sample_size:
+                    skipped += 1
+                    self.skipped_indices.add(i)
+                    continue
+                self.fnames.append(self.text_compressor.compress(items[0]))
+                sizes.append(sz)
+        logger.info(f"loaded {len(self.fnames)}, skipped {skipped} samples")
+
+        self.sizes = np.array(sizes, dtype=np.int64)
+
+        try:
+            import pyarrow
+
+            self.fnames = pyarrow.array(self.fnames)
+        except:
+            logger.debug(
+                "Could not create a pyarrow array. Please install pyarrow for better performance"
+            )
+            pass
+
+        self.set_bucket_info(num_buckets)
+
+    def __getitem__(self, index):
+        import soundfile as sf
+
+        fn = self.fnames[index]
+        fn = fn if isinstance(self.fnames, list) else fn.as_py()
+        fn = self.text_compressor.decompress(fn)
+        path_or_fp = os.path.join(self.root_dir, fn)
+        _path, slice_ptr = parse_path(path_or_fp)
+        if len(slice_ptr) == 2:
+            byte_data = read_from_stored_zip(_path, slice_ptr[0], slice_ptr[1])
+            assert is_sf_audio_data(byte_data)
+            path_or_fp = io.BytesIO(byte_data)
+
+        wav, curr_sample_rate = sf.read(path_or_fp, dtype="float32")
+
+        feats = torch.from_numpy(wav).float()
+        feats = self.postprocess(feats, curr_sample_rate)
+        return {"id": index, "source": feats}
+
+
+class BinarizedAudioDataset(RawAudioDataset):
+    def __init__(
+        self,
+        data_dir,
+        split,
+        sample_rate,
+        max_sample_size=None,
+        min_sample_size=0,
+        shuffle=True,
+        pad=False,
+        normalize=False,
+        num_buckets=0,
+        compute_mask_indices=False,
+        **mask_compute_kwargs,
+    ):
+        super().__init__(
+            sample_rate=sample_rate,
+            max_sample_size=max_sample_size,
+            min_sample_size=min_sample_size,
+            shuffle=shuffle,
+            pad=pad,
+            normalize=normalize,
+            compute_mask_indices=compute_mask_indices,
+            **mask_compute_kwargs,
+        )
+
+        from fairseq.data import data_utils, Dictionary
+
+        self.fnames_dict = Dictionary.load(os.path.join(data_dir, "dict.txt"))
+
+        root_path = os.path.join(data_dir, f"{split}.root")
+        if os.path.exists(root_path):
+            with open(root_path, "r") as f:
+                self.root_dir = next(f).strip()
+        else:
+            self.root_dir = None
+
+        fnames_path = os.path.join(data_dir, split)
+        self.fnames = data_utils.load_indexed_dataset(fnames_path, self.fnames_dict)
+        lengths_path = os.path.join(data_dir, f"{split}.lengths")
+
+        with open(lengths_path, "r") as f:
+            for line in f:
+                sz = int(line.rstrip())
+                assert (
+                    sz >= min_sample_size
+                ), f"Min sample size is not supported for binarized dataset, but found a sample with size {sz}"
+                self.sizes.append(sz)
+
+        self.sizes = np.array(self.sizes, dtype=np.int64)
+
+        self.set_bucket_info(num_buckets)
+        logger.info(f"loaded {len(self.fnames)} samples")
+
+    def __getitem__(self, index):
+        import soundfile as sf
+
+        fname = self.fnames_dict.string(self.fnames[index], separator="")
+        if self.root_dir:
+            fname = os.path.join(self.root_dir, fname)
+
+        wav, curr_sample_rate = sf.read(fname)
+        feats = torch.from_numpy(wav).float()
+        feats = self.postprocess(feats, curr_sample_rate)
+        return {"id": index, "source": feats}

modules/voice_conversion/fairseq/data/audio/speech_to_speech_dataset.py

@@ -0,0 +1,379 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+import logging
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Dict, List, Optional, Tuple
+
+import torch
+
+from fairseq.data import ConcatDataset, Dictionary
+from fairseq.data import data_utils as fairseq_data_utils
+from fairseq.data.audio.audio_utils import get_features_or_waveform
+from fairseq.data.audio.data_cfg import S2SDataConfig
+from fairseq.data.audio.speech_to_text_dataset import (
+    SpeechToTextDataset,
+    SpeechToTextDatasetCreator,
+    TextTargetMultitaskData,
+    _collate_frames,
+)
+
+logger = logging.getLogger(__name__)
+
+
+@dataclass
+class SpeechToSpeechDatasetItem(object):
+    index: int
+    source: torch.Tensor
+    target: Optional[torch.Tensor] = None
+    target_speaker: Optional[torch.Tensor] = None
+    tgt_lang_tag: Optional[int] = None
+
+
+class SpeechToSpeechDataset(SpeechToTextDataset):
+    def __init__(
+        self,
+        split: str,
+        is_train_split: bool,
+        data_cfg: S2SDataConfig,
+        src_audio_paths: List[str],
+        src_n_frames: List[int],
+        tgt_audio_paths: List[str],
+        tgt_n_frames: List[int],
+        src_langs: Optional[List[str]] = None,
+        tgt_langs: Optional[List[str]] = None,
+        ids: Optional[List[str]] = None,
+        target_is_code: bool = False,
+        tgt_dict: Dictionary = None,
+        n_frames_per_step: int = 1,
+    ):
+        tgt_texts = tgt_audio_paths if target_is_code else None
+        super().__init__(
+            split=split,
+            is_train_split=is_train_split,
+            cfg=data_cfg,
+            audio_paths=src_audio_paths,
+            n_frames=src_n_frames,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            tgt_texts=tgt_texts,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            n_frames_per_step=n_frames_per_step,
+        )
+
+        self.tgt_audio_paths = tgt_audio_paths
+        self.tgt_lens = [t // self.n_frames_per_step for t in tgt_n_frames]
+
+        assert not target_is_code or tgt_dict is not None
+        self.target_is_code = target_is_code
+
+        assert len(tgt_audio_paths) == self.n_samples
+        assert len(tgt_n_frames) == self.n_samples
+
+        self.tgt_speakers = None
+        if self.cfg.target_speaker_embed:
+            samples = SpeechToTextDatasetCreator._load_samples_from_tsv(
+                self.cfg.target_speaker_embed, split
+            )
+            spk_emb_dict = {s["id"]: s["speaker_embed"] for s in samples}
+            self.tgt_speakers = [spk_emb_dict[id] for id in self.ids]
+            assert len(self.tgt_speakers) == self.n_samples
+
+        logger.info(self.__repr__())
+
+    def pack_units(self, input: torch.Tensor) -> torch.Tensor:
+        if self.n_frames_per_step <= 1:
+            return input
+
+        offset = 4
+        vocab_size = (
+            len(self.tgt_dict) - offset
+        )  # remove offset from <bos>, <pad>, <eos>, <unk>, which is specific to fairseq dictionary
+
+        assert input.dim() == 1
+        stacked_input = (
+            input[:-1].view(-1, self.n_frames_per_step) - offset
+        )  # remove <eos>
+        scale = [
+            pow(vocab_size, self.n_frames_per_step - 1 - i)
+            for i in range(self.n_frames_per_step)
+        ]
+        scale = torch.LongTensor(scale).squeeze(0)
+        res = input.new((len(input) - 1) // self.n_frames_per_step + 1).fill_(input[-1])
+        res[:-1] = (stacked_input * scale).sum(dim=1) + offset
+
+        return res
+
+    def __getitem__(self, index: int) -> SpeechToSpeechDatasetItem:
+        source = self._get_source_audio(index)
+
+        tgt_lang_tag = None
+        if self.cfg.prepend_tgt_lang_tag_as_bos:
+            # prepend_tgt_lang_tag_as_bos: put tgt_lang_tag as bos of target
+            tgt_lang_tag = self.get_lang_tag_idx(self.tgt_langs[index], self.tgt_dict)
+
+        if not self.target_is_code:
+            target = get_features_or_waveform(self.tgt_audio_paths[index])
+            target = torch.from_numpy(target).float()
+            target = self.pack_frames(target)
+        else:
+            target = self.tgt_dict.encode_line(
+                self.tgt_audio_paths[index],
+                add_if_not_exist=False,
+                append_eos=True,
+            ).long()
+            if self.n_frames_per_step > 1:
+                n_tgt_frame = target.size(0) - 1  # exclude <eos>
+                keep_n_tgt_frame = n_tgt_frame - n_tgt_frame % self.n_frames_per_step
+                target = torch.cat(
+                    (
+                        target[:keep_n_tgt_frame],
+                        target.new_full((1,), self.tgt_dict.eos()),
+                    ),
+                    dim=0,
+                )
+
+        if self.tgt_speakers:
+            tgt_spk = get_features_or_waveform(self.tgt_speakers[index])
+            tgt_spk = torch.from_numpy(tgt_spk).float()
+        else:
+            tgt_spk = torch.FloatTensor([])
+
+        return SpeechToSpeechDatasetItem(
+            index=index,
+            source=source,
+            target=target,
+            target_speaker=tgt_spk,
+            tgt_lang_tag=tgt_lang_tag,
+        )
+
+    def _collate_target(self, samples: List[SpeechToSpeechDatasetItem]) -> torch.Tensor:
+        if self.target_is_code:
+            target = fairseq_data_utils.collate_tokens(
+                [x.target for x in samples],
+                self.tgt_dict.pad(),
+                self.tgt_dict.eos(),
+                left_pad=False,
+                move_eos_to_beginning=False,
+            )
+            # convert stacked units to a single id
+            pack_targets = [self.pack_units(x.target) for x in samples]
+            prev_output_tokens = fairseq_data_utils.collate_tokens(
+                pack_targets,
+                self.tgt_dict.pad(),
+                self.tgt_dict.eos(),
+                left_pad=False,
+                move_eos_to_beginning=True,
+            )
+            target_lengths = torch.tensor(
+                [x.size(0) for x in pack_targets], dtype=torch.long
+            )
+        else:
+            target = _collate_frames([x.target for x in samples], is_audio_input=False)
+            bsz, _, d = target.size()
+            prev_output_tokens = torch.cat(
+                (target.new_full((bsz, 1, d), 0.0), target[:, :-1, :]), dim=1
+            )
+            target_lengths = torch.tensor(
+                [x.target.size(0) for x in samples], dtype=torch.long
+            )
+
+        return target, prev_output_tokens, target_lengths
+
+    def collater(
+        self, samples: List[SpeechToSpeechDatasetItem], return_order: bool = False
+    ) -> Dict:
+        if len(samples) == 0:
+            return {}
+        indices = torch.tensor([x.index for x in samples], dtype=torch.long)
+        frames = _collate_frames([x.source for x in samples], self.cfg.use_audio_input)
+        # sort samples by descending number of frames
+        n_frames = torch.tensor([x.source.size(0) for x in samples], dtype=torch.long)
+        n_frames, order = n_frames.sort(descending=True)
+        indices = indices.index_select(0, order)
+        frames = frames.index_select(0, order)
+
+        target, prev_output_tokens, target_lengths = self._collate_target(samples)
+        target = target.index_select(0, order)
+        target_lengths = target_lengths.index_select(0, order)
+        prev_output_tokens = prev_output_tokens.index_select(0, order)
+        ntokens = sum(x.target.size(0) for x in samples)
+
+        tgt_speakers = None
+        if self.cfg.target_speaker_embed:
+            tgt_speakers = _collate_frames(
+                [x.target_speaker for x in samples], is_audio_input=True
+            ).index_select(0, order)
+
+        net_input = {
+            "src_tokens": frames,
+            "src_lengths": n_frames,
+            "prev_output_tokens": prev_output_tokens,
+            "tgt_speaker": tgt_speakers,  # TODO: unify "speaker" and "tgt_speaker"
+        }
+        if self.tgt_texts is not None and samples[0].tgt_lang_tag is not None:
+            for i in range(len(samples)):
+                net_input["prev_output_tokens"][i][0] = samples[order[i]].tgt_lang_tag
+        out = {
+            "id": indices,
+            "net_input": net_input,
+            "speaker": tgt_speakers,  # to support Tacotron2 loss for speech-to-spectrogram model
+            "target": target,
+            "target_lengths": target_lengths,
+            "ntokens": ntokens,
+            "nsentences": len(samples),
+        }
+        if return_order:
+            out["order"] = order
+        return out
+
+
+class SpeechToSpeechMultitaskDataset(SpeechToSpeechDataset):
+    def __init__(self, **kwargs):
+        super().__init__(**kwargs)
+        self.multitask_data = {}
+
+    def add_multitask_dataset(self, task_name, task_data):
+        self.multitask_data[task_name] = task_data
+
+    def __getitem__(
+        self, index: int
+    ) -> Tuple[SpeechToSpeechDatasetItem, Dict[str, torch.Tensor]]:
+        s2s_data = super().__getitem__(index)
+
+        multitask_target = {}
+        sample_id = self.ids[index]
+        tgt_lang = self.tgt_langs[index]
+        for task_name, task_dataset in self.multitask_data.items():
+            multitask_target[task_name] = task_dataset.get(sample_id, tgt_lang)
+
+        return s2s_data, multitask_target
+
+    def collater(
+        self, samples: List[Tuple[SpeechToSpeechDatasetItem, Dict[str, torch.Tensor]]]
+    ) -> Dict:
+        if len(samples) == 0:
+            return {}
+
+        out = super().collater([s for s, _ in samples], return_order=True)
+        order = out["order"]
+        del out["order"]
+
+        for task_name, task_dataset in self.multitask_data.items():
+            if "multitask" not in out:
+                out["multitask"] = {}
+            d = [s[task_name] for _, s in samples]
+            task_target = task_dataset.collater(d)
+            out["multitask"][task_name] = {
+                "target": task_target["target"].index_select(0, order),
+                "target_lengths": task_target["target_lengths"].index_select(0, order),
+                "ntokens": task_target["ntokens"],
+            }
+            out["multitask"][task_name]["net_input"] = {
+                "prev_output_tokens": task_target["prev_output_tokens"].index_select(
+                    0, order
+                ),
+            }
+
+        return out
+
+
+class SpeechToSpeechDatasetCreator(object):
+    # mandatory columns
+    KEY_ID, KEY_SRC_AUDIO, KEY_SRC_N_FRAMES = "id", "src_audio", "src_n_frames"
+    KEY_TGT_AUDIO, KEY_TGT_N_FRAMES = "tgt_audio", "tgt_n_frames"
+    # optional columns
+    KEY_SRC_LANG, KEY_TGT_LANG = "src_lang", "tgt_lang"
+    # default values
+    DEFAULT_LANG = ""
+
+    @classmethod
+    def _from_list(
+        cls,
+        split_name: str,
+        is_train_split,
+        samples: List[Dict],
+        data_cfg: S2SDataConfig,
+        target_is_code: bool = False,
+        tgt_dict: Dictionary = None,
+        n_frames_per_step: int = 1,
+        multitask: Optional[Dict] = None,
+    ) -> SpeechToSpeechDataset:
+        audio_root = Path(data_cfg.audio_root)
+        ids = [s[cls.KEY_ID] for s in samples]
+        src_audio_paths = [
+            (audio_root / s[cls.KEY_SRC_AUDIO]).as_posix() for s in samples
+        ]
+        tgt_audio_paths = [
+            s[cls.KEY_TGT_AUDIO]
+            if target_is_code
+            else (audio_root / s[cls.KEY_TGT_AUDIO]).as_posix()
+            for s in samples
+        ]
+        src_n_frames = [int(s[cls.KEY_SRC_N_FRAMES]) for s in samples]
+        tgt_n_frames = [int(s[cls.KEY_TGT_N_FRAMES]) for s in samples]
+        src_langs = [s.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for s in samples]
+        tgt_langs = [s.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for s in samples]
+
+        has_multitask = multitask is not None and len(multitask.keys()) > 0
+        dataset_cls = (
+            SpeechToSpeechMultitaskDataset if has_multitask else SpeechToSpeechDataset
+        )
+
+        ds = dataset_cls(
+            split=split_name,
+            is_train_split=is_train_split,
+            data_cfg=data_cfg,
+            src_audio_paths=src_audio_paths,
+            src_n_frames=src_n_frames,
+            tgt_audio_paths=tgt_audio_paths,
+            tgt_n_frames=tgt_n_frames,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            target_is_code=target_is_code,
+            tgt_dict=tgt_dict,
+            n_frames_per_step=n_frames_per_step,
+        )
+
+        if has_multitask:
+            for task_name, task_obj in multitask.items():
+                task_data = TextTargetMultitaskData(
+                    task_obj.args, split_name, task_obj.target_dictionary
+                )
+                ds.add_multitask_dataset(task_name, task_data)
+        return ds
+
+    @classmethod
+    def from_tsv(
+        cls,
+        root: str,
+        data_cfg: S2SDataConfig,
+        splits: str,
+        is_train_split: bool,
+        epoch: int,
+        seed: int,
+        target_is_code: bool = False,
+        tgt_dict: Dictionary = None,
+        n_frames_per_step: int = 1,
+        multitask: Optional[Dict] = None,
+    ) -> SpeechToSpeechDataset:
+        datasets = []
+        for split in splits.split(","):
+            samples = SpeechToTextDatasetCreator._load_samples_from_tsv(root, split)
+            ds = cls._from_list(
+                split_name=split,
+                is_train_split=is_train_split,
+                samples=samples,
+                data_cfg=data_cfg,
+                target_is_code=target_is_code,
+                tgt_dict=tgt_dict,
+                n_frames_per_step=n_frames_per_step,
+                multitask=multitask,
+            )
+            datasets.append(ds)
+        return ConcatDataset(datasets) if len(datasets) > 1 else datasets[0]

modules/voice_conversion/fairseq/data/audio/speech_to_text_dataset.py

@@ -0,0 +1,733 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+import csv
+import logging
+import re
+from argparse import Namespace
+from collections import defaultdict
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Dict, List, Optional, Tuple, Union
+
+import numpy as np
+import torch
+import torch.nn.functional as F
+
+from fairseq.data import ConcatDataset, Dictionary, FairseqDataset, ResamplingDataset
+from fairseq.data import data_utils as fairseq_data_utils
+from fairseq.data import encoders
+from fairseq.data.audio.audio_utils import get_features_or_waveform
+from fairseq.data.audio.data_cfg import S2TDataConfig
+from fairseq.data.audio.dataset_transforms import CompositeAudioDatasetTransform
+from fairseq.data.audio.dataset_transforms.concataugment import ConcatAugment
+from fairseq.data.audio.dataset_transforms.noisyoverlapaugment import (
+    NoisyOverlapAugment,
+)
+from fairseq.data.audio.feature_transforms import CompositeAudioFeatureTransform
+from fairseq.data.audio.waveform_transforms import CompositeAudioWaveformTransform
+
+logger = logging.getLogger(__name__)
+
+
+def _collate_frames(
+    frames: List[torch.Tensor], is_audio_input: bool = False
+) -> torch.Tensor:
+    """
+    Convert a list of 2D frames into a padded 3D tensor
+    Args:
+        frames (list): list of 2D frames of size L[i]*f_dim. Where L[i] is
+            length of i-th frame and f_dim is static dimension of features
+    Returns:
+        3D tensor of size len(frames)*len_max*f_dim where len_max is max of L[i]
+    """
+    max_len = max(frame.size(0) for frame in frames)
+    if is_audio_input:
+        out = frames[0].new_zeros((len(frames), max_len))
+    else:
+        out = frames[0].new_zeros((len(frames), max_len, frames[0].size(1)))
+    for i, v in enumerate(frames):
+        out[i, : v.size(0)] = v
+    return out
+
+
+def _is_int_or_np_int(n):
+    return isinstance(n, int) or (
+        isinstance(n, np.generic) and isinstance(n.item(), int)
+    )
+
+
+@dataclass
+class SpeechToTextDatasetItem(object):
+    index: int
+    source: torch.Tensor
+    target: Optional[torch.Tensor] = None
+    speaker_id: Optional[int] = None
+
+
+class SpeechToTextDataset(FairseqDataset):
+    LANG_TAG_TEMPLATE = "<lang:{}>"
+
+    def __init__(
+        self,
+        split: str,
+        is_train_split: bool,
+        cfg: S2TDataConfig,
+        audio_paths: List[str],
+        n_frames: List[int],
+        src_texts: Optional[List[str]] = None,
+        tgt_texts: Optional[List[str]] = None,
+        speakers: Optional[List[str]] = None,
+        src_langs: Optional[List[str]] = None,
+        tgt_langs: Optional[List[str]] = None,
+        ids: Optional[List[str]] = None,
+        tgt_dict: Optional[Dictionary] = None,
+        pre_tokenizer=None,
+        bpe_tokenizer=None,
+        n_frames_per_step=1,
+        speaker_to_id=None,
+        append_eos=True,
+    ):
+        self.split, self.is_train_split = split, is_train_split
+        self.cfg = cfg
+        self.audio_paths, self.n_frames = audio_paths, n_frames
+        self.n_samples = len(audio_paths)
+        assert len(n_frames) == self.n_samples > 0
+        assert src_texts is None or len(src_texts) == self.n_samples
+        assert tgt_texts is None or len(tgt_texts) == self.n_samples
+        assert speakers is None or len(speakers) == self.n_samples
+        assert src_langs is None or len(src_langs) == self.n_samples
+        assert tgt_langs is None or len(tgt_langs) == self.n_samples
+        assert ids is None or len(ids) == self.n_samples
+        assert (tgt_dict is None and tgt_texts is None) or (
+            tgt_dict is not None and tgt_texts is not None
+        )
+        self.src_texts, self.tgt_texts = src_texts, tgt_texts
+        self.src_langs, self.tgt_langs = src_langs, tgt_langs
+        self.speakers = speakers
+        self.tgt_dict = tgt_dict
+        self.check_tgt_lang_tag()
+        self.ids = ids
+        self.shuffle = cfg.shuffle if is_train_split else False
+
+        self.feature_transforms = CompositeAudioFeatureTransform.from_config_dict(
+            self.cfg.get_feature_transforms(split, is_train_split)
+        )
+        self.waveform_transforms = CompositeAudioWaveformTransform.from_config_dict(
+            self.cfg.get_waveform_transforms(split, is_train_split)
+        )
+        # TODO: add these to data_cfg.py
+        self.dataset_transforms = CompositeAudioDatasetTransform.from_config_dict(
+            self.cfg.get_dataset_transforms(split, is_train_split)
+        )
+
+        # check proper usage of transforms
+        if self.feature_transforms and self.cfg.use_audio_input:
+            logger.warning(
+                "Feature transforms will not be applied. To use feature transforms, "
+                "set use_audio_input as False in config."
+            )
+
+        self.pre_tokenizer = pre_tokenizer
+        self.bpe_tokenizer = bpe_tokenizer
+        self.n_frames_per_step = n_frames_per_step
+        self.speaker_to_id = speaker_to_id
+
+        self.tgt_lens = self.get_tgt_lens_and_check_oov()
+        self.append_eos = append_eos
+
+        logger.info(self.__repr__())
+
+    def get_tgt_lens_and_check_oov(self):
+        if self.tgt_texts is None:
+            return [0 for _ in range(self.n_samples)]
+        tgt_lens = []
+        n_tokens, n_oov_tokens = 0, 0
+        for i in range(self.n_samples):
+            tokenized = self.get_tokenized_tgt_text(i).split(" ")
+            oov_tokens = [
+                t
+                for t in tokenized
+                if self.tgt_dict.index(t) == self.tgt_dict.unk_index
+            ]
+            n_tokens += len(tokenized)
+            n_oov_tokens += len(oov_tokens)
+            tgt_lens.append(len(tokenized))
+        logger.info(f"'{self.split}' has {n_oov_tokens / n_tokens * 100:.2f}% OOV")
+        return tgt_lens
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + f'(split="{self.split}", n_samples={self.n_samples:_}, '
+            f"prepend_tgt_lang_tag={self.cfg.prepend_tgt_lang_tag}, "
+            f"n_frames_per_step={self.n_frames_per_step}, "
+            f"shuffle={self.shuffle}, "
+            f"feature_transforms={self.feature_transforms}, "
+            f"waveform_transforms={self.waveform_transforms}, "
+            f"dataset_transforms={self.dataset_transforms})"
+        )
+
+    @classmethod
+    def is_lang_tag(cls, token):
+        pattern = cls.LANG_TAG_TEMPLATE.replace("{}", "(.*)")
+        return re.match(pattern, token)
+
+    def check_tgt_lang_tag(self):
+        if self.cfg.prepend_tgt_lang_tag:
+            assert self.tgt_langs is not None and self.tgt_dict is not None
+            tgt_lang_tags = [
+                self.LANG_TAG_TEMPLATE.format(t) for t in set(self.tgt_langs)
+            ]
+            assert all(t in self.tgt_dict for t in tgt_lang_tags)
+
+    @classmethod
+    def tokenize(cls, tokenizer, text: str):
+        return text if tokenizer is None else tokenizer.encode(text)
+
+    def get_tokenized_tgt_text(self, index: Union[int, List[int]]):
+        if _is_int_or_np_int(index):
+            text = self.tgt_texts[index]
+        else:
+            text = " ".join([self.tgt_texts[i] for i in index])
+
+        text = self.tokenize(self.pre_tokenizer, text)
+        text = self.tokenize(self.bpe_tokenizer, text)
+        return text
+
+    def pack_frames(self, feature: torch.Tensor):
+        if self.n_frames_per_step == 1:
+            return feature
+        n_packed_frames = feature.shape[0] // self.n_frames_per_step
+        feature = feature[: self.n_frames_per_step * n_packed_frames]
+        return feature.reshape(n_packed_frames, -1)
+
+    @classmethod
+    def get_lang_tag_idx(cls, lang: str, dictionary: Dictionary):
+        lang_tag_idx = dictionary.index(cls.LANG_TAG_TEMPLATE.format(lang))
+        assert lang_tag_idx != dictionary.unk()
+        return lang_tag_idx
+
+    def _get_source_audio(self, index: Union[int, List[int]]) -> torch.Tensor:
+        """
+        Gives source audio for given index with any relevant transforms
+        applied. For ConcatAug, source audios for given indices are
+        concatenated in given order.
+        Args:
+            index (int or List[int]): index—or in the case of ConcatAug,
+            indices—to pull the source audio for
+        Returns:
+            source audios concatenated for given indices with
+            relevant transforms appplied
+        """
+        if _is_int_or_np_int(index):
+            source = get_features_or_waveform(
+                self.audio_paths[index],
+                need_waveform=self.cfg.use_audio_input,
+                use_sample_rate=self.cfg.use_sample_rate,
+                waveform_transforms=self.waveform_transforms,
+            )
+        else:
+            source = np.concatenate(
+                [
+                    get_features_or_waveform(
+                        self.audio_paths[i],
+                        need_waveform=self.cfg.use_audio_input,
+                        use_sample_rate=self.cfg.use_sample_rate,
+                        waveform_transforms=self.waveform_transforms,
+                    )
+                    for i in index
+                ]
+            )
+        if self.cfg.use_audio_input:
+            source = torch.from_numpy(source).float()
+            if self.cfg.standardize_audio:
+                with torch.no_grad():
+                    source = F.layer_norm(source, source.shape)
+        else:
+            if self.feature_transforms is not None:
+                source = self.feature_transforms(source)
+            source = torch.from_numpy(source).float()
+        return source
+
+    def __getitem__(self, index: int) -> SpeechToTextDatasetItem:
+        has_concat = self.dataset_transforms.has_transform(ConcatAugment)
+        if has_concat:
+            concat = self.dataset_transforms.get_transform(ConcatAugment)
+            indices = concat.find_indices(index, self.n_frames, self.n_samples)
+
+        source = self._get_source_audio(indices if has_concat else index)
+        source = self.pack_frames(source)
+
+        target = None
+        if self.tgt_texts is not None:
+            tokenized = self.get_tokenized_tgt_text(indices if has_concat else index)
+            target = self.tgt_dict.encode_line(
+                tokenized, add_if_not_exist=False, append_eos=self.append_eos
+            ).long()
+            if self.cfg.prepend_tgt_lang_tag:
+                lang_tag_idx = self.get_lang_tag_idx(
+                    self.tgt_langs[index], self.tgt_dict
+                )
+                target = torch.cat((torch.LongTensor([lang_tag_idx]), target), 0)
+
+        if self.cfg.prepend_bos_and_append_tgt_lang_tag:
+            bos = torch.LongTensor([self.tgt_dict.bos()])
+            lang_tag_idx = self.get_lang_tag_idx(self.tgt_langs[index], self.tgt_dict)
+            assert lang_tag_idx != self.tgt_dict.unk()
+            lang_tag_idx = torch.LongTensor([lang_tag_idx])
+            target = torch.cat((bos, target, lang_tag_idx), 0)
+
+        speaker_id = None
+        if self.speaker_to_id is not None:
+            speaker_id = self.speaker_to_id[self.speakers[index]]
+        return SpeechToTextDatasetItem(
+            index=index, source=source, target=target, speaker_id=speaker_id
+        )
+
+    def __len__(self):
+        return self.n_samples
+
+    def collater(
+        self, samples: List[SpeechToTextDatasetItem], return_order: bool = False
+    ) -> Dict:
+        if len(samples) == 0:
+            return {}
+        indices = torch.tensor([x.index for x in samples], dtype=torch.long)
+
+        sources = [x.source for x in samples]
+        has_NOAug = self.dataset_transforms.has_transform(NoisyOverlapAugment)
+        if has_NOAug and self.cfg.use_audio_input:
+            NOAug = self.dataset_transforms.get_transform(NoisyOverlapAugment)
+            sources = NOAug(sources)
+
+        frames = _collate_frames(sources, self.cfg.use_audio_input)
+        # sort samples by descending number of frames
+        n_frames = torch.tensor([x.size(0) for x in sources], dtype=torch.long)
+        n_frames, order = n_frames.sort(descending=True)
+        indices = indices.index_select(0, order)
+        frames = frames.index_select(0, order)
+
+        target, target_lengths = None, None
+        prev_output_tokens = None
+        ntokens = None
+        if self.tgt_texts is not None:
+            target = fairseq_data_utils.collate_tokens(
+                [x.target for x in samples],
+                self.tgt_dict.pad(),
+                self.tgt_dict.eos(),
+                left_pad=False,
+                move_eos_to_beginning=False,
+            )
+            target = target.index_select(0, order)
+            target_lengths = torch.tensor(
+                [x.target.size(0) for x in samples], dtype=torch.long
+            ).index_select(0, order)
+            prev_output_tokens = fairseq_data_utils.collate_tokens(
+                [x.target for x in samples],
+                self.tgt_dict.pad(),
+                eos_idx=None,
+                left_pad=False,
+                move_eos_to_beginning=True,
+            )
+            prev_output_tokens = prev_output_tokens.index_select(0, order)
+            ntokens = sum(x.target.size(0) for x in samples)
+
+        speaker = None
+        if self.speaker_to_id is not None:
+            speaker = (
+                torch.tensor([s.speaker_id for s in samples], dtype=torch.long)
+                .index_select(0, order)
+                .view(-1, 1)
+            )
+
+        net_input = {
+            "src_tokens": frames,
+            "src_lengths": n_frames,
+            "prev_output_tokens": prev_output_tokens,
+        }
+        out = {
+            "id": indices,
+            "net_input": net_input,
+            "speaker": speaker,
+            "target": target,
+            "target_lengths": target_lengths,
+            "ntokens": ntokens,
+            "nsentences": len(samples),
+        }
+        if return_order:
+            out["order"] = order
+        return out
+
+    def num_tokens(self, index):
+        return self.n_frames[index]
+
+    def size(self, index):
+        return self.n_frames[index], self.tgt_lens[index]
+
+    @property
+    def sizes(self):
+        return np.array(self.n_frames)
+
+    @property
+    def can_reuse_epoch_itr_across_epochs(self):
+        return True
+
+    def ordered_indices(self):
+        if self.shuffle:
+            order = [np.random.permutation(len(self))]
+        else:
+            order = [np.arange(len(self))]
+        # first by descending order of # of frames then by original/random order
+        order.append([-n for n in self.n_frames])
+        return np.lexsort(order)
+
+    def prefetch(self, indices):
+        raise False
+
+
+class TextTargetMultitaskData(object):
+    # mandatory columns
+    KEY_ID, KEY_TEXT = "id", "tgt_text"
+    LANG_TAG_TEMPLATE = "<lang:{}>"
+
+    def __init__(self, args, split, tgt_dict):
+        samples = SpeechToTextDatasetCreator._load_samples_from_tsv(args.data, split)
+        self.data = {s[self.KEY_ID]: s[self.KEY_TEXT] for s in samples}
+        self.dict = tgt_dict
+        self.append_eos = args.decoder_type != "ctc"
+        self.pre_tokenizer = self.build_tokenizer(args)
+        self.bpe_tokenizer = self.build_bpe(args)
+        self.prepend_bos_and_append_tgt_lang_tag = (
+            args.prepend_bos_and_append_tgt_lang_tag
+        )
+        self.eos_token = args.eos_token
+        self.lang_tag_mapping = args.get_lang_tag_mapping
+
+    @classmethod
+    def is_lang_tag(cls, token):
+        pattern = cls.LANG_TAG_TEMPLATE.replace("{}", "(.*)")
+        return re.match(pattern, token)
+
+    @classmethod
+    def tokenize(cls, tokenizer, text: str):
+        return text if tokenizer is None else tokenizer.encode(text)
+
+    def get_tokenized_tgt_text(self, index: int):
+        text = self.tokenize(self.pre_tokenizer, self.data[index])
+        text = self.tokenize(self.bpe_tokenizer, text)
+        return text
+
+    def get_lang_tag_idx(self, lang: str, dictionary: Dictionary):
+        lang_tag = self.LANG_TAG_TEMPLATE.format(lang)
+        lang_tag = self.lang_tag_mapping.get(lang_tag, lang_tag)
+        lang_tag_idx = dictionary.index(lang_tag)
+        assert lang_tag_idx != dictionary.unk(), (lang, lang_tag)
+        return lang_tag_idx
+
+    def build_tokenizer(self, args):
+        pre_tokenizer = args.config.get("pre_tokenizer")
+        if pre_tokenizer is not None:
+            logger.info(f"pre-tokenizer: {pre_tokenizer}")
+            return encoders.build_tokenizer(Namespace(**pre_tokenizer))
+        else:
+            return None
+
+    def build_bpe(self, args):
+        bpe_tokenizer = args.config.get("bpe_tokenizer")
+        if bpe_tokenizer is not None:
+            logger.info(f"tokenizer: {bpe_tokenizer}")
+            return encoders.build_bpe(Namespace(**bpe_tokenizer))
+        else:
+            return None
+
+    def get(self, sample_id, tgt_lang=None):
+        if sample_id in self.data:
+            tokenized = self.get_tokenized_tgt_text(sample_id)
+            target = self.dict.encode_line(
+                tokenized,
+                add_if_not_exist=False,
+                append_eos=self.append_eos,
+            )
+            if self.prepend_bos_and_append_tgt_lang_tag:
+                bos = torch.LongTensor([self.dict.bos()])
+                lang_tag_idx = self.get_lang_tag_idx(tgt_lang, self.dict)
+                assert lang_tag_idx != self.dict.unk()
+                lang_tag_idx = torch.LongTensor([lang_tag_idx])
+                target = torch.cat((bos, target, lang_tag_idx), 0)
+            return target
+        else:
+            logger.warning(f"no target for {sample_id}")
+            return torch.IntTensor([])
+
+    def collater(self, samples: List[torch.Tensor]) -> torch.Tensor:
+        out = fairseq_data_utils.collate_tokens(
+            samples,
+            self.dict.pad(),
+            eos_idx=None,
+            left_pad=False,
+            move_eos_to_beginning=False,
+        ).long()
+
+        prev_out = fairseq_data_utils.collate_tokens(
+            samples,
+            self.dict.pad(),
+            eos_idx=None,
+            left_pad=False,
+            move_eos_to_beginning=True,
+        ).long()
+
+        target_lengths = torch.tensor([t.size(0) for t in samples], dtype=torch.long)
+        ntokens = sum(t.size(0) for t in samples)
+
+        output = {
+            "prev_output_tokens": prev_out,
+            "target": out,
+            "target_lengths": target_lengths,
+            "ntokens": ntokens,
+        }
+
+        return output
+
+
+class SpeechToTextMultitaskDataset(SpeechToTextDataset):
+    def __init__(self, **kwargs):
+        super().__init__(**kwargs)
+        self.multitask_data = {}
+
+    def add_multitask_dataset(self, task_name, task_data):
+        self.multitask_data[task_name] = task_data
+
+    def __getitem__(
+        self, index: int
+    ) -> Tuple[SpeechToTextDatasetItem, Dict[str, torch.Tensor]]:
+        s2t_data = super().__getitem__(index)
+
+        multitask_target = {}
+        sample_id = self.ids[index]
+        tgt_lang = self.tgt_langs[index]
+        for task_name, task_dataset in self.multitask_data.items():
+            multitask_target[task_name] = task_dataset.get(sample_id, tgt_lang)
+
+        return s2t_data, multitask_target
+
+    def collater(
+        self, samples: List[Tuple[SpeechToTextDatasetItem, Dict[str, torch.Tensor]]]
+    ) -> Dict:
+        if len(samples) == 0:
+            return {}
+
+        out = super().collater([s for s, _ in samples], return_order=True)
+        order = out["order"]
+        del out["order"]
+
+        for task_name, task_dataset in self.multitask_data.items():
+            if "multitask" not in out:
+                out["multitask"] = {}
+            d = [s[task_name] for _, s in samples]
+            task_target = task_dataset.collater(d)
+            out["multitask"][task_name] = {
+                "target": task_target["target"].index_select(0, order),
+                "target_lengths": task_target["target_lengths"].index_select(0, order),
+                "ntokens": task_target["ntokens"],
+            }
+            out["multitask"][task_name]["net_input"] = {
+                "prev_output_tokens": task_target["prev_output_tokens"].index_select(
+                    0, order
+                ),
+            }
+
+        return out
+
+
+class SpeechToTextDatasetCreator(object):
+    # mandatory columns
+    KEY_ID, KEY_AUDIO, KEY_N_FRAMES = "id", "audio", "n_frames"
+    KEY_TGT_TEXT = "tgt_text"
+    # optional columns
+    KEY_SPEAKER, KEY_SRC_TEXT = "speaker", "src_text"
+    KEY_SRC_LANG, KEY_TGT_LANG = "src_lang", "tgt_lang"
+    # default values
+    DEFAULT_SPEAKER = DEFAULT_SRC_TEXT = DEFAULT_LANG = ""
+
+    @classmethod
+    def _from_list(
+        cls,
+        split_name: str,
+        is_train_split,
+        samples: List[Dict],
+        cfg: S2TDataConfig,
+        tgt_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        n_frames_per_step,
+        speaker_to_id,
+        multitask: Optional[Dict] = None,
+    ) -> SpeechToTextDataset:
+        audio_root = Path(cfg.audio_root)
+        ids = [s[cls.KEY_ID] for s in samples]
+        audio_paths = [(audio_root / s[cls.KEY_AUDIO]).as_posix() for s in samples]
+        n_frames = [int(s[cls.KEY_N_FRAMES]) for s in samples]
+        tgt_texts = [s[cls.KEY_TGT_TEXT] for s in samples]
+        src_texts = [s.get(cls.KEY_SRC_TEXT, cls.DEFAULT_SRC_TEXT) for s in samples]
+        speakers = [s.get(cls.KEY_SPEAKER, cls.DEFAULT_SPEAKER) for s in samples]
+        src_langs = [s.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for s in samples]
+        tgt_langs = [s.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for s in samples]
+
+        has_multitask = multitask is not None and len(multitask.keys()) > 0
+        dataset_cls = (
+            SpeechToTextMultitaskDataset if has_multitask else SpeechToTextDataset
+        )
+
+        ds = dataset_cls(
+            split=split_name,
+            is_train_split=is_train_split,
+            cfg=cfg,
+            audio_paths=audio_paths,
+            n_frames=n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            n_frames_per_step=n_frames_per_step,
+            speaker_to_id=speaker_to_id,
+        )
+
+        if has_multitask:
+            for task_name, task_obj in multitask.items():
+                task_data = TextTargetMultitaskData(
+                    task_obj.args, split_name, task_obj.target_dictionary
+                )
+                ds.add_multitask_dataset(task_name, task_data)
+        return ds
+
+    @classmethod
+    def get_size_ratios(
+        cls, datasets: List[SpeechToTextDataset], alpha: float = 1.0
+    ) -> List[float]:
+        """Size ratios for temperature-based sampling
+        (https://arxiv.org/abs/1907.05019)"""
+
+        id_to_lp, lp_to_sz = {}, defaultdict(int)
+        for ds in datasets:
+            lang_pairs = {f"{s}->{t}" for s, t in zip(ds.src_langs, ds.tgt_langs)}
+            assert len(lang_pairs) == 1
+            lang_pair = list(lang_pairs)[0]
+            id_to_lp[ds.split] = lang_pair
+            lp_to_sz[lang_pair] += sum(ds.n_frames)
+
+        sz_sum = sum(v for v in lp_to_sz.values())
+        lp_to_prob = {k: v / sz_sum for k, v in lp_to_sz.items()}
+        lp_to_tgt_prob = {k: v**alpha for k, v in lp_to_prob.items()}
+        prob_sum = sum(v for v in lp_to_tgt_prob.values())
+        lp_to_tgt_prob = {k: v / prob_sum for k, v in lp_to_tgt_prob.items()}
+        lp_to_sz_ratio = {
+            k: (lp_to_tgt_prob[k] * sz_sum) / v for k, v in lp_to_sz.items()
+        }
+        size_ratio = [lp_to_sz_ratio[id_to_lp[ds.split]] for ds in datasets]
+
+        p_formatted = {
+            k: f"{lp_to_prob[k]:.3f}->{lp_to_tgt_prob[k]:.3f}" for k in lp_to_sz
+        }
+        logger.info(f"sampling probability balancing: {p_formatted}")
+        sr_formatted = {ds.split: f"{r:.3f}" for ds, r in zip(datasets, size_ratio)}
+        logger.info(f"balanced sampling size ratio: {sr_formatted}")
+        return size_ratio
+
+    @classmethod
+    def _load_samples_from_tsv(cls, root: str, split: str):
+        tsv_path = Path(root) / f"{split}.tsv"
+        if not tsv_path.is_file():
+            raise FileNotFoundError(f"Dataset not found: {tsv_path}")
+        with open(tsv_path) as f:
+            reader = csv.DictReader(
+                f,
+                delimiter="\t",
+                quotechar=None,
+                doublequote=False,
+                lineterminator="\n",
+                quoting=csv.QUOTE_NONE,
+            )
+            samples = [dict(e) for e in reader]
+        if len(samples) == 0:
+            raise ValueError(f"Empty manifest: {tsv_path}")
+        return samples
+
+    @classmethod
+    def _from_tsv(
+        cls,
+        root: str,
+        cfg: S2TDataConfig,
+        split: str,
+        tgt_dict,
+        is_train_split: bool,
+        pre_tokenizer,
+        bpe_tokenizer,
+        n_frames_per_step,
+        speaker_to_id,
+        multitask: Optional[Dict] = None,
+    ) -> SpeechToTextDataset:
+        samples = cls._load_samples_from_tsv(root, split)
+        return cls._from_list(
+            split,
+            is_train_split,
+            samples,
+            cfg,
+            tgt_dict,
+            pre_tokenizer,
+            bpe_tokenizer,
+            n_frames_per_step,
+            speaker_to_id,
+            multitask,
+        )
+
+    @classmethod
+    def from_tsv(
+        cls,
+        root: str,
+        cfg: S2TDataConfig,
+        splits: str,
+        tgt_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        is_train_split: bool,
+        epoch: int,
+        seed: int,
+        n_frames_per_step: int = 1,
+        speaker_to_id=None,
+        multitask: Optional[Dict] = None,
+    ) -> SpeechToTextDataset:
+        datasets = [
+            cls._from_tsv(
+                root=root,
+                cfg=cfg,
+                split=split,
+                tgt_dict=tgt_dict,
+                is_train_split=is_train_split,
+                pre_tokenizer=pre_tokenizer,
+                bpe_tokenizer=bpe_tokenizer,
+                n_frames_per_step=n_frames_per_step,
+                speaker_to_id=speaker_to_id,
+                multitask=multitask,
+            )
+            for split in splits.split(",")
+        ]
+
+        if is_train_split and len(datasets) > 1 and cfg.sampling_alpha != 1.0:
+            # temperature-based sampling
+            size_ratios = cls.get_size_ratios(datasets, alpha=cfg.sampling_alpha)
+            datasets = [
+                ResamplingDataset(
+                    d, size_ratio=r, seed=seed, epoch=epoch, replace=(r >= 1.0)
+                )
+                for r, d in zip(size_ratios, datasets)
+            ]
+
+        return ConcatDataset(datasets) if len(datasets) > 1 else datasets[0]

modules/voice_conversion/fairseq/data/audio/speech_to_text_joint_dataset.py

@@ -0,0 +1,359 @@
+# Copyright (c) Facebook, Inc. and its affiliates.
+#
+# This source code is licensed under the MIT license found in the
+# LICENSE file in the root directory of this source tree.
+
+import logging
+from pathlib import Path
+from typing import Dict, List, NamedTuple, Optional
+
+import torch
+
+from fairseq.data import ConcatDataset, Dictionary, ResamplingDataset
+from fairseq.data import data_utils as fairseq_data_utils
+from fairseq.data.audio.speech_to_text_dataset import (
+    S2TDataConfig,
+    SpeechToTextDataset,
+    SpeechToTextDatasetCreator,
+)
+
+logger = logging.getLogger(__name__)
+
+
+class S2TJointDataConfig(S2TDataConfig):
+    """Wrapper class for data config YAML"""
+
+    @property
+    def src_vocab_filename(self):
+        """fairseq vocabulary file under data root"""
+        return self.config.get("src_vocab_filename", "src_dict.txt")
+
+    @property
+    def src_pre_tokenizer(self) -> Dict:
+        """Pre-tokenizer to apply before subword tokenization. Returning
+        a dictionary with `tokenizer` providing the tokenizer name and
+        the other items providing the tokenizer-specific arguments.
+        Tokenizers are defined in `fairseq.data.encoders.*`"""
+        return self.config.get("src_pre_tokenizer", {"tokenizer": None})
+
+    @property
+    def src_bpe_tokenizer(self) -> Dict:
+        """Subword tokenizer to apply on source text after pre-tokenization.
+        Returning a dictionary with `bpe` providing the tokenizer name and
+        the other items providing the tokenizer-specific arguments.
+        Tokenizers are defined in `fairseq.data.encoders.*`"""
+        return self.config.get("src_bpe_tokenizer", {"bpe": None})
+
+    @property
+    def prepend_tgt_lang_tag_no_change(self) -> bool:
+        """Prepend target lang ID token as the prev_output_tokens BOS (e.g. for
+        to-many multilingual setting). No change needed during inference.
+        This option is deprecated and replaced by prepend_tgt_lang_tag_as_bos.
+        """
+        value = self.config.get("prepend_tgt_lang_tag_no_change", None)
+        if value is None:
+            return self.config.get("prepend_tgt_lang_tag_as_bos", False)
+        return value
+
+    @property
+    def sampling_text_alpha(self):
+        """Hyper-parameter alpha = 1/T for temperature-based resampling. (text
+        input only) (alpha = 1 for no resampling)"""
+        return self.config.get("sampling_text_alpha", 1.0)
+
+
+class SpeechToTextJointDatasetItem(NamedTuple):
+    index: int
+    source: torch.Tensor
+    target: Optional[torch.Tensor] = None
+    src_txt_tokens: Optional[torch.Tensor] = None
+    tgt_lang_tag: Optional[int] = None
+    src_lang_tag: Optional[int] = None
+    tgt_alignment: Optional[torch.Tensor] = None
+
+
+# use_src_lang_id:
+#   0: don't use src_lang_id
+#   1: attach src_lang_id to the src_txt_tokens as eos
+class SpeechToTextJointDataset(SpeechToTextDataset):
+    def __init__(
+        self,
+        split: str,
+        is_train_split: bool,
+        cfg: S2TJointDataConfig,
+        audio_paths: List[str],
+        n_frames: List[int],
+        src_texts: Optional[List[str]] = None,
+        tgt_texts: Optional[List[str]] = None,
+        speakers: Optional[List[str]] = None,
+        src_langs: Optional[List[str]] = None,
+        tgt_langs: Optional[List[str]] = None,
+        ids: Optional[List[str]] = None,
+        tgt_dict: Optional[Dictionary] = None,
+        src_dict: Optional[Dictionary] = None,
+        pre_tokenizer=None,
+        bpe_tokenizer=None,
+        src_pre_tokenizer=None,
+        src_bpe_tokenizer=None,
+        append_eos: Optional[bool] = True,
+        alignment: Optional[List[str]] = None,
+        use_src_lang_id: Optional[int] = 0,
+    ):
+        super().__init__(
+            split,
+            is_train_split,
+            cfg,
+            audio_paths,
+            n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            append_eos=append_eos,
+        )
+
+        self.src_dict = src_dict
+        self.src_pre_tokenizer = src_pre_tokenizer
+        self.src_bpe_tokenizer = src_bpe_tokenizer
+        self.alignment = None
+        self.use_src_lang_id = use_src_lang_id
+        if alignment is not None:
+            self.alignment = [
+                [float(s) for s in sample.split()] for sample in alignment
+            ]
+
+    def get_tokenized_src_text(self, index: int):
+        text = self.tokenize(self.src_pre_tokenizer, self.src_texts[index])
+        text = self.tokenize(self.src_bpe_tokenizer, text)
+        return text
+
+    def __getitem__(self, index: int) -> SpeechToTextJointDatasetItem:
+        s2t_dataset_item = super().__getitem__(index)
+        src_tokens = None
+        src_lang_tag = None
+        if self.src_texts is not None and self.src_dict is not None:
+            src_tokens = self.get_tokenized_src_text(index)
+            src_tokens = self.src_dict.encode_line(
+                src_tokens, add_if_not_exist=False, append_eos=True
+            ).long()
+            if self.use_src_lang_id > 0:
+                src_lang_tag = self.get_lang_tag_idx(
+                    self.src_langs[index], self.src_dict
+                )
+        tgt_lang_tag = None
+        if self.cfg.prepend_tgt_lang_tag_no_change:
+            # prepend_tgt_lang_tag_no_change: modify prev_output_tokens instead
+            tgt_lang_tag = self.get_lang_tag_idx(self.tgt_langs[index], self.tgt_dict)
+        ali = None
+        if self.alignment is not None:
+            ali = torch.Tensor(self.alignment[index]).float()
+
+        return SpeechToTextJointDatasetItem(
+            index=index,
+            source=s2t_dataset_item.source,
+            target=s2t_dataset_item.target,
+            src_txt_tokens=src_tokens,
+            tgt_lang_tag=tgt_lang_tag,
+            src_lang_tag=src_lang_tag,
+            tgt_alignment=ali,
+        )
+
+    def __len__(self):
+        return self.n_samples
+
+    def collater(self, samples: List[SpeechToTextJointDatasetItem]) -> Dict:
+        s2t_out = super().collater(samples, return_order=True)
+        if s2t_out == {}:
+            return s2t_out
+        net_input, order = s2t_out["net_input"], s2t_out["order"]
+
+        if self.src_texts is not None and self.src_dict is not None:
+            src_txt_tokens = fairseq_data_utils.collate_tokens(
+                [x.src_txt_tokens for x in samples],
+                self.src_dict.pad(),
+                self.src_dict.eos(),
+                left_pad=False,
+                move_eos_to_beginning=False,
+            )
+            src_txt_lengths = torch.tensor(
+                [x.src_txt_tokens.size()[0] for x in samples], dtype=torch.long
+            )
+            if self.use_src_lang_id > 0:
+                src_lang_idxs = torch.tensor(
+                    [s.src_lang_tag for s in samples], dtype=src_txt_tokens.dtype
+                )
+                if self.use_src_lang_id == 1:  # replace eos with lang_id
+                    eos_idx = src_txt_lengths - 1
+                    src_txt_tokens.scatter_(
+                        1, eos_idx.view(-1, 1), src_lang_idxs.view(-1, 1)
+                    )
+                else:
+                    raise NotImplementedError("Implementation is required")
+
+            src_txt_tokens = src_txt_tokens.index_select(0, order)
+            src_txt_lengths = src_txt_lengths.index_select(0, order)
+            net_input["src_txt_tokens"] = src_txt_tokens
+            net_input["src_txt_lengths"] = src_txt_lengths
+
+        net_input["alignment"] = None
+        if self.alignment is not None:
+            max_len = max([s.tgt_alignment.size(0) for s in samples])
+            alignment = torch.ones(len(samples), max_len).float()
+            for i, s in enumerate(samples):
+                cur_len = s.tgt_alignment.size(0)
+                alignment[i][:cur_len].copy_(s.tgt_alignment)
+            net_input["alignment"] = alignment.index_select(0, order)
+
+        if self.tgt_texts is not None and samples[0].tgt_lang_tag is not None:
+            for i in range(len(samples)):
+                net_input["prev_output_tokens"][i][0] = samples[order[i]].tgt_lang_tag
+
+        out = {
+            "id": s2t_out["id"],
+            "net_input": net_input,
+            "target": s2t_out["target"],
+            "target_lengths": s2t_out["target_lengths"],
+            "ntokens": s2t_out["ntokens"],
+            "nsentences": len(samples),
+        }
+        return out
+
+
+class SpeechToTextJointDatasetCreator(SpeechToTextDatasetCreator):
+    KEY_ALIGN = "align"
+
+    @classmethod
+    def _from_list(
+        cls,
+        split_name: str,
+        is_train_split,
+        samples: List[Dict],
+        cfg: S2TJointDataConfig,
+        tgt_dict,
+        src_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        src_pre_tokenizer,
+        src_bpe_tokenizer,
+        append_eos,
+        use_src_lang_id,
+    ) -> SpeechToTextJointDataset:
+        audio_root = Path(cfg.audio_root)
+        ids = [s[cls.KEY_ID] for s in samples]
+        audio_paths = [(audio_root / s[cls.KEY_AUDIO]).as_posix() for s in samples]
+        n_frames = [int(s[cls.KEY_N_FRAMES]) for s in samples]
+        tgt_texts = [s[cls.KEY_TGT_TEXT] for s in samples]
+        src_texts = [s.get(cls.KEY_SRC_TEXT, cls.DEFAULT_SRC_TEXT) for s in samples]
+        speakers = [s.get(cls.KEY_SPEAKER, cls.DEFAULT_SPEAKER) for s in samples]
+        src_langs = [s.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for s in samples]
+        tgt_langs = [s.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for s in samples]
+        tgt_alignment = None
+        if cls.KEY_ALIGN in samples[0].keys():
+            tgt_alignment = [s[cls.KEY_ALIGN] for s in samples]
+        return SpeechToTextJointDataset(
+            split_name,
+            is_train_split,
+            cfg,
+            audio_paths,
+            n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            src_dict=src_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            src_pre_tokenizer=src_pre_tokenizer,
+            src_bpe_tokenizer=src_bpe_tokenizer,
+            append_eos=append_eos,
+            alignment=tgt_alignment,
+            use_src_lang_id=use_src_lang_id,
+        )
+
+    @classmethod
+    def _from_tsv(
+        cls,
+        root: str,
+        cfg: S2TJointDataConfig,
+        split: str,
+        tgt_dict,
+        src_dict,
+        is_train_split: bool,
+        pre_tokenizer,
+        bpe_tokenizer,
+        src_pre_tokenizer,
+        src_bpe_tokenizer,
+        append_eos: bool,
+        use_src_lang_id: int,
+    ) -> SpeechToTextJointDataset:
+        samples = cls._load_samples_from_tsv(root, split)
+        return cls._from_list(
+            split,
+            is_train_split,
+            samples,
+            cfg,
+            tgt_dict,
+            src_dict,
+            pre_tokenizer,
+            bpe_tokenizer,
+            src_pre_tokenizer,
+            src_bpe_tokenizer,
+            append_eos,
+            use_src_lang_id,
+        )
+
+    @classmethod
+    def from_tsv(
+        cls,
+        root: str,
+        cfg: S2TJointDataConfig,
+        splits: str,
+        tgt_dict,
+        src_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        src_pre_tokenizer,
+        src_bpe_tokenizer,
+        is_train_split: bool,
+        epoch: int,
+        seed: int,
+        append_eos: Optional[bool] = True,
+        use_src_lang_id: Optional[int] = 0,
+    ) -> SpeechToTextJointDataset:
+        datasets = [
+            cls._from_tsv(
+                root,
+                cfg,
+                split,
+                tgt_dict,
+                src_dict,
+                is_train_split,
+                pre_tokenizer,
+                bpe_tokenizer,
+                src_pre_tokenizer,
+                src_bpe_tokenizer,
+                append_eos=append_eos,
+                use_src_lang_id=use_src_lang_id,
+            )
+            for split in splits.split(",")
+        ]
+
+        if is_train_split and len(datasets) > 1 and cfg.sampling_alpha != 1.0:
+            # temperature-based sampling
+            size_ratios = cls.get_size_ratios(datasets, alpha=cfg.sampling_alpha)
+            datasets = [
+                ResamplingDataset(
+                    d, size_ratio=r, seed=seed, epoch=epoch, replace=(r >= 1.0)
+                )
+                for r, d in zip(size_ratios, datasets)
+            ]
+
+        return ConcatDataset(datasets) if len(datasets) > 1 else datasets[0]

modules/voice_conversion/fairseq/data/audio/text_to_speech_dataset.py

@@ -0,0 +1,250 @@
+# Copyright (c) 2017-present, Facebook, Inc.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the LICENSE file in
+# the root directory of this source tree. An additional grant of patent rights
+# can be found in the PATENTS file in the same directory.abs
+
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Any, Dict, List, Optional
+
+import numpy as np
+import torch
+
+from fairseq.data import Dictionary
+from fairseq.data import data_utils as fairseq_data_utils
+from fairseq.data.audio.audio_utils import get_features_or_waveform
+from fairseq.data.audio.speech_to_text_dataset import (
+    S2TDataConfig,
+    SpeechToTextDataset,
+    SpeechToTextDatasetCreator,
+    _collate_frames,
+)
+
+
+@dataclass
+class TextToSpeechDatasetItem(object):
+    index: int
+    source: torch.Tensor
+    target: Optional[torch.Tensor] = None
+    speaker_id: Optional[int] = None
+    duration: Optional[torch.Tensor] = None
+    pitch: Optional[torch.Tensor] = None
+    energy: Optional[torch.Tensor] = None
+
+
+class TextToSpeechDataset(SpeechToTextDataset):
+    def __init__(
+        self,
+        split: str,
+        is_train_split: bool,
+        cfg: S2TDataConfig,
+        audio_paths: List[str],
+        n_frames: List[int],
+        src_texts: Optional[List[str]] = None,
+        tgt_texts: Optional[List[str]] = None,
+        speakers: Optional[List[str]] = None,
+        src_langs: Optional[List[str]] = None,
+        tgt_langs: Optional[List[str]] = None,
+        ids: Optional[List[str]] = None,
+        tgt_dict: Optional[Dictionary] = None,
+        pre_tokenizer=None,
+        bpe_tokenizer=None,
+        n_frames_per_step=1,
+        speaker_to_id=None,
+        durations: Optional[List[List[int]]] = None,
+        pitches: Optional[List[str]] = None,
+        energies: Optional[List[str]] = None,
+    ):
+        super(TextToSpeechDataset, self).__init__(
+            split,
+            is_train_split,
+            cfg,
+            audio_paths,
+            n_frames,
+            src_texts=src_texts,
+            tgt_texts=tgt_texts,
+            speakers=speakers,
+            src_langs=src_langs,
+            tgt_langs=tgt_langs,
+            ids=ids,
+            tgt_dict=tgt_dict,
+            pre_tokenizer=pre_tokenizer,
+            bpe_tokenizer=bpe_tokenizer,
+            n_frames_per_step=n_frames_per_step,
+            speaker_to_id=speaker_to_id,
+        )
+        self.durations = durations
+        self.pitches = pitches
+        self.energies = energies
+
+    def __getitem__(self, index: int) -> TextToSpeechDatasetItem:
+        s2t_item = super().__getitem__(index)
+
+        duration, pitch, energy = None, None, None
+        if self.durations is not None:
+            duration = torch.tensor(
+                self.durations[index] + [0], dtype=torch.long  # pad 0 for EOS
+            )
+        if self.pitches is not None:
+            pitch = get_features_or_waveform(self.pitches[index])
+            pitch = torch.from_numpy(
+                np.concatenate((pitch, [0]))  # pad 0 for EOS
+            ).float()
+        if self.energies is not None:
+            energy = get_features_or_waveform(self.energies[index])
+            energy = torch.from_numpy(
+                np.concatenate((energy, [0]))  # pad 0 for EOS
+            ).float()
+        return TextToSpeechDatasetItem(
+            index=index,
+            source=s2t_item.source,
+            target=s2t_item.target,
+            speaker_id=s2t_item.speaker_id,
+            duration=duration,
+            pitch=pitch,
+            energy=energy,
+        )
+
+    def collater(self, samples: List[TextToSpeechDatasetItem]) -> Dict[str, Any]:
+        if len(samples) == 0:
+            return {}
+
+        src_lengths, order = torch.tensor(
+            [s.target.shape[0] for s in samples], dtype=torch.long
+        ).sort(descending=True)
+        id_ = torch.tensor([s.index for s in samples], dtype=torch.long).index_select(
+            0, order
+        )
+        feat = _collate_frames(
+            [s.source for s in samples], self.cfg.use_audio_input
+        ).index_select(0, order)
+        target_lengths = torch.tensor(
+            [s.source.shape[0] for s in samples], dtype=torch.long
+        ).index_select(0, order)
+
+        src_tokens = fairseq_data_utils.collate_tokens(
+            [s.target for s in samples],
+            self.tgt_dict.pad(),
+            self.tgt_dict.eos(),
+            left_pad=False,
+            move_eos_to_beginning=False,
+        ).index_select(0, order)
+
+        speaker = None
+        if self.speaker_to_id is not None:
+            speaker = (
+                torch.tensor([s.speaker_id for s in samples], dtype=torch.long)
+                .index_select(0, order)
+                .view(-1, 1)
+            )
+
+        bsz, _, d = feat.size()
+        prev_output_tokens = torch.cat(
+            (feat.new_zeros((bsz, 1, d)), feat[:, :-1, :]), dim=1
+        )
+
+        durations, pitches, energies = None, None, None
+        if self.durations is not None:
+            durations = fairseq_data_utils.collate_tokens(
+                [s.duration for s in samples], 0
+            ).index_select(0, order)
+            assert src_tokens.shape[1] == durations.shape[1]
+        if self.pitches is not None:
+            pitches = _collate_frames([s.pitch for s in samples], True)
+            pitches = pitches.index_select(0, order)
+            assert src_tokens.shape[1] == pitches.shape[1]
+        if self.energies is not None:
+            energies = _collate_frames([s.energy for s in samples], True)
+            energies = energies.index_select(0, order)
+            assert src_tokens.shape[1] == energies.shape[1]
+        src_texts = [self.tgt_dict.string(samples[i].target) for i in order]
+
+        return {
+            "id": id_,
+            "net_input": {
+                "src_tokens": src_tokens,
+                "src_lengths": src_lengths,
+                "prev_output_tokens": prev_output_tokens,
+            },
+            "speaker": speaker,
+            "target": feat,
+            "durations": durations,
+            "pitches": pitches,
+            "energies": energies,
+            "target_lengths": target_lengths,
+            "ntokens": sum(target_lengths).item(),
+            "nsentences": len(samples),
+            "src_texts": src_texts,
+        }
+
+
+class TextToSpeechDatasetCreator(SpeechToTextDatasetCreator):
+    KEY_DURATION = "duration"
+    KEY_PITCH = "pitch"
+    KEY_ENERGY = "energy"
+
+    @classmethod
+    def _from_list(
+        cls,
+        split_name: str,
+        is_train_split,
+        samples: List[Dict],
+        cfg: S2TDataConfig,
+        tgt_dict,
+        pre_tokenizer,
+        bpe_tokenizer,
+        n_frames_per_step,
+        speaker_to_id,
+        multitask=None,
+    ) -> TextToSpeechDataset:
+        audio_root = Path(cfg.audio_root)
+        ids = [s[cls.KEY_ID] for s in samples]
+        audio_paths = [(audio_root / s[cls.KEY_AUDIO]).as_posix() for s in samples]
+        n_frames = [int(s[cls.KEY_N_FRAMES]) for s in samples]
+        tgt_texts = [s[cls.KEY_TGT_TEXT] for s in samples]
+        src_texts = [s.get(cls.KEY_SRC_TEXT, cls.DEFAULT_SRC_TEXT) for s in samples]
+        speakers = [s.get(cls.KEY_SPEAKER, cls.DEFAULT_SPEAKER) for s in samples]
+        src_langs = [s.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for s in samples]
+        tgt_langs = [s.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for s in samples]
+
+        durations = [s.get(cls.KEY_DURATION, None) for s in samples]
+        durations = [
+            None if dd is None else [int(d) for d in dd.split(" ")] for dd in durations
+        ]
+        durations = None if any(dd is None for dd in durations) else durations
+
+        pitches = [s.get(cls.KEY_PITCH, None) for s in samples]
+        pitches = [
+            None if pp is None else (audio_root / pp).as_posix() for pp in pitches
+        ]
+        pitches = None if any(pp is None for pp in pitches) else pitches
+
+        energies = [s.get(cls.KEY_ENERGY, None) for s in samples]
+        energies = [
+            None if ee is None else (audio_root / ee).as_posix() for ee in energies
+        ]
+        energies = None if any(ee is None for ee in energies) else energies
+
+        return TextToSpeechDataset(
+            split_name,
+            is_train_split,
+            cfg,
+            audio_paths,
+            n_frames,
+            src_texts,
+            tgt_texts,
+            speakers,
+            src_langs,
+            tgt_langs,
+            ids,
+            tgt_dict,
+            pre_tokenizer,
+            bpe_tokenizer,
+            n_frames_per_step,
+            speaker_to_id,
+            durations,
+            pitches,
+            energies,
+        )

modules/voice_conversion/fairseq/data/audio/waveform_transforms/__init__.py

@@ -0,0 +1,48 @@
+import os
+from fairseq.data.audio import (
+    AudioTransform,
+    CompositeAudioTransform,
+    import_transforms,
+    register_audio_transform,
+)
+
+
+class AudioWaveformTransform(AudioTransform):
+    pass
+
+
+AUDIO_WAVEFORM_TRANSFORM_REGISTRY = {}
+AUDIO_WAVEFORM_TRANSFORM_CLASS_NAMES = set()
+
+
+def get_audio_waveform_transform(name):
+    return AUDIO_WAVEFORM_TRANSFORM_REGISTRY[name]
+
+
+def register_audio_waveform_transform(name):
+    return register_audio_transform(
+        name,
+        AudioWaveformTransform,
+        AUDIO_WAVEFORM_TRANSFORM_REGISTRY,
+        AUDIO_WAVEFORM_TRANSFORM_CLASS_NAMES,
+    )
+
+
+import_transforms(os.path.dirname(__file__), "waveform")
+
+
+class CompositeAudioWaveformTransform(CompositeAudioTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        return super()._from_config_dict(
+            cls,
+            "waveform",
+            get_audio_waveform_transform,
+            CompositeAudioWaveformTransform,
+            config,
+        )
+
+    def __call__(self, x, sample_rate):
+        for t in self.transforms:
+            x, sample_rate = t(x, sample_rate)
+        return x, sample_rate

modules/voice_conversion/fairseq/data/audio/waveform_transforms/noiseaugment.py

@@ -0,0 +1,201 @@
+from pathlib import Path
+import numpy as np
+from math import ceil
+
+from fairseq.data.audio import rand_uniform
+from fairseq.data.audio.waveform_transforms import (
+    AudioWaveformTransform,
+    register_audio_waveform_transform,
+)
+
+SNR_MIN = 5.0
+SNR_MAX = 15.0
+RATE = 0.25
+
+NOISE_RATE = 1.0
+NOISE_LEN_MEAN = 0.2
+NOISE_LEN_STD = 0.05
+
+
+class NoiseAugmentTransform(AudioWaveformTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return cls(
+            _config.get("samples_path", None),
+            _config.get("snr_min", SNR_MIN),
+            _config.get("snr_max", SNR_MAX),
+            _config.get("rate", RATE),
+        )
+
+    def __init__(
+        self,
+        samples_path: str,
+        snr_min: float = SNR_MIN,
+        snr_max: float = SNR_MAX,
+        rate: float = RATE,
+    ):
+        # Sanity checks
+        assert (
+            samples_path
+        ), "need to provide path to audio samples for noise augmentation"
+        assert snr_max >= snr_min, f"empty signal-to-noise range ({snr_min}, {snr_max})"
+        assert rate >= 0 and rate <= 1, "rate should be a float between 0 to 1"
+
+        self.paths = list(Path(samples_path).glob("**/*.wav"))  # load music
+        self.n_samples = len(self.paths)
+        assert self.n_samples > 0, f"no audio files found in {samples_path}"
+
+        self.snr_min = snr_min
+        self.snr_max = snr_max
+        self.rate = rate
+
+    def __repr__(self):
+        return (
+            self.__class__.__name__
+            + "("
+            + ", ".join(
+                [
+                    f"n_samples={self.n_samples}",
+                    f"snr={self.snr_min}-{self.snr_max}dB",
+                    f"rate={self.rate}",
+                ]
+            )
+            + ")"
+        )
+
+    def pick_sample(self, goal_shape, always_2d=False, use_sample_rate=None):
+        from fairseq.data.audio.audio_utils import get_waveform
+
+        path = self.paths[np.random.randint(0, self.n_samples)]
+        sample = get_waveform(
+            path, always_2d=always_2d, output_sample_rate=use_sample_rate
+        )[0]
+
+        # Check dimensions match, else silently skip adding noise to sample
+        # NOTE: SHOULD THIS QUIT WITH AN ERROR?
+        is_2d = len(goal_shape) == 2
+        if len(goal_shape) != sample.ndim or (
+            is_2d and goal_shape[0] != sample.shape[0]
+        ):
+            return np.zeros(goal_shape)
+
+        # Cut/repeat sample to size
+        len_dim = len(goal_shape) - 1
+        n_repeat = ceil(goal_shape[len_dim] / sample.shape[len_dim])
+        repeated = np.tile(sample, [1, n_repeat] if is_2d else n_repeat)
+        start = np.random.randint(0, repeated.shape[len_dim] - goal_shape[len_dim] + 1)
+        return (
+            repeated[:, start : start + goal_shape[len_dim]]
+            if is_2d
+            else repeated[start : start + goal_shape[len_dim]]
+        )
+
+    def _mix(self, source, noise, snr):
+        get_power = lambda x: np.mean(x**2)
+        if get_power(noise):
+            scl = np.sqrt(
+                get_power(source) / (np.power(10, snr / 10) * get_power(noise))
+            )
+        else:
+            scl = 0
+        return 1 * source + scl * noise
+
+    def _get_noise(self, goal_shape, always_2d=False, use_sample_rate=None):
+        return self.pick_sample(goal_shape, always_2d, use_sample_rate)
+
+    def __call__(self, source, sample_rate):
+        if np.random.random() > self.rate:
+            return source, sample_rate
+
+        noise = self._get_noise(
+            source.shape, always_2d=True, use_sample_rate=sample_rate
+        )
+
+        return (
+            self._mix(source, noise, rand_uniform(self.snr_min, self.snr_max)),
+            sample_rate,
+        )
+
+
+@register_audio_waveform_transform("musicaugment")
+class MusicAugmentTransform(NoiseAugmentTransform):
+    pass
+
+
+@register_audio_waveform_transform("backgroundnoiseaugment")
+class BackgroundNoiseAugmentTransform(NoiseAugmentTransform):
+    pass
+
+
+@register_audio_waveform_transform("babbleaugment")
+class BabbleAugmentTransform(NoiseAugmentTransform):
+    def _get_noise(self, goal_shape, always_2d=False, use_sample_rate=None):
+        for i in range(np.random.randint(3, 8)):
+            speech = self.pick_sample(goal_shape, always_2d, use_sample_rate)
+            if i == 0:
+                agg_noise = speech
+            else:  # SNR scaled by i (how many noise signals already in agg_noise)
+                agg_noise = self._mix(agg_noise, speech, i)
+        return agg_noise
+
+
+@register_audio_waveform_transform("sporadicnoiseaugment")
+class SporadicNoiseAugmentTransform(NoiseAugmentTransform):
+    @classmethod
+    def from_config_dict(cls, config=None):
+        _config = {} if config is None else config
+        return cls(
+            _config.get("samples_path", None),
+            _config.get("snr_min", SNR_MIN),
+            _config.get("snr_max", SNR_MAX),
+            _config.get("rate", RATE),
+            _config.get("noise_rate", NOISE_RATE),
+            _config.get("noise_len_mean", NOISE_LEN_MEAN),
+            _config.get("noise_len_std", NOISE_LEN_STD),
+        )
+
+    def __init__(
+        self,
+        samples_path: str,
+        snr_min: float = SNR_MIN,
+        snr_max: float = SNR_MAX,
+        rate: float = RATE,
+        noise_rate: float = NOISE_RATE,  # noises per second
+        noise_len_mean: float = NOISE_LEN_MEAN,  # length of noises in seconds
+        noise_len_std: float = NOISE_LEN_STD,
+    ):
+        super().__init__(samples_path, snr_min, snr_max, rate)
+        self.noise_rate = noise_rate
+        self.noise_len_mean = noise_len_mean
+        self.noise_len_std = noise_len_std
+
+    def _get_noise(self, goal_shape, always_2d=False, use_sample_rate=None):
+        agg_noise = np.zeros(goal_shape)
+        len_dim = len(goal_shape) - 1
+        is_2d = len(goal_shape) == 2
+
+        n_noises = round(self.noise_rate * goal_shape[len_dim] / use_sample_rate)
+        start_pointers = [
+            round(rand_uniform(0, goal_shape[len_dim])) for _ in range(n_noises)
+        ]
+
+        for start_pointer in start_pointers:
+            noise_shape = list(goal_shape)
+            len_seconds = np.random.normal(self.noise_len_mean, self.noise_len_std)
+            noise_shape[len_dim] = round(max(0, len_seconds) * use_sample_rate)
+            end_pointer = start_pointer + noise_shape[len_dim]
+            if end_pointer >= goal_shape[len_dim]:
+                continue
+
+            noise = self.pick_sample(noise_shape, always_2d, use_sample_rate)
+            if is_2d:
+                agg_noise[:, start_pointer:end_pointer] = (
+                    agg_noise[:, start_pointer:end_pointer] + noise
+                )
+            else:
+                agg_noise[start_pointer:end_pointer] = (
+                    agg_noise[start_pointer:end_pointer] + noise
+                )
+
+        return agg_noise