# Copyright (c) 2024 NVIDIA CORPORATION. # Licensed under the MIT license. # Adapted from https://github.com/jik876/hifi-gan under the MIT license. # LICENSE is in incl_licenses directory. import math import os import random import torch import torch.utils.data import numpy as np from librosa.util import normalize from scipy.io.wavfile import read from librosa.filters import mel as librosa_mel_fn import pathlib from tqdm import tqdm MAX_WAV_VALUE = 32767.0 # NOTE: 32768.0 -1 to prevent int16 overflow (results in popping sound in corner cases) def load_wav(full_path, sr_target): sampling_rate, data = read(full_path) if sampling_rate != sr_target: raise RuntimeError( f"Sampling rate of the file {full_path} is {sampling_rate} Hz, but the model requires {sr_target} Hz" ) return data, sampling_rate def dynamic_range_compression(x, C=1, clip_val=1e-5): return np.log(np.clip(x, a_min=clip_val, a_max=None) * C) def dynamic_range_decompression(x, C=1): return np.exp(x) / C def dynamic_range_compression_torch(x, C=1, clip_val=1e-5): return torch.log(torch.clamp(x, min=clip_val) * C) def dynamic_range_decompression_torch(x, C=1): return torch.exp(x) / C def spectral_normalize_torch(magnitudes): return dynamic_range_compression_torch(magnitudes) def spectral_de_normalize_torch(magnitudes): return dynamic_range_decompression_torch(magnitudes) mel_basis_cache = {} hann_window_cache = {} def mel_spectrogram( y: torch.Tensor, n_fft: int, num_mels: int, sampling_rate: int, hop_size: int, win_size: int, fmin: int, fmax: int = None, center: bool = False, ) -> torch.Tensor: """ Calculate the mel spectrogram of an input signal. This function uses slaney norm for the librosa mel filterbank (using librosa.filters.mel) and uses Hann window for STFT (using torch.stft). Args: y (torch.Tensor): Input signal. n_fft (int): FFT size. num_mels (int): Number of mel bins. sampling_rate (int): Sampling rate of the input signal. hop_size (int): Hop size for STFT. win_size (int): Window size for STFT. fmin (int): Minimum frequency for mel filterbank. fmax (int): Maximum frequency for mel filterbank. If None, defaults to half the sampling rate (fmax = sr / 2.0) inside librosa_mel_fn center (bool): Whether to pad the input to center the frames. Default is False. Returns: torch.Tensor: Mel spectrogram. """ if torch.min(y) < -1.0: print(f"[WARNING] Min value of input waveform signal is {torch.min(y)}") if torch.max(y) > 1.0: print(f"[WARNING] Max value of input waveform signal is {torch.max(y)}") device = y.device key = f"{n_fft}_{num_mels}_{sampling_rate}_{hop_size}_{win_size}_{fmin}_{fmax}_{device}" if key not in mel_basis_cache: mel = librosa_mel_fn( sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax ) mel_basis_cache[key] = torch.from_numpy(mel).float().to(device) hann_window_cache[key] = torch.hann_window(win_size).to(device) mel_basis = mel_basis_cache[key] hann_window = hann_window_cache[key] padding = (n_fft - hop_size) // 2 y = torch.nn.functional.pad( y.unsqueeze(1), (padding, padding), mode="reflect" ).squeeze(1) spec = torch.stft( y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window, center=center, pad_mode="reflect", normalized=False, onesided=True, return_complex=True, ) spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9) mel_spec = torch.matmul(mel_basis, spec) mel_spec = spectral_normalize_torch(mel_spec) return mel_spec def get_mel_spectrogram(wav, h): """ Generate mel spectrogram from a waveform using given hyperparameters. Args: wav (torch.Tensor): Input waveform. h: Hyperparameters object with attributes n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax. Returns: torch.Tensor: Mel spectrogram. """ return mel_spectrogram( wav, h.n_fft, h.num_mels, h.sampling_rate, h.hop_size, h.win_size, h.fmin, h.fmax, ) def get_dataset_filelist(a): training_files = [] validation_files = [] list_unseen_validation_files = [] with open(a.input_training_file, "r", encoding="utf-8") as fi: training_files = [ os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav") for x in fi.read().split("\n") if len(x) > 0 ] print(f"first training file: {training_files[0]}") with open(a.input_validation_file, "r", encoding="utf-8") as fi: validation_files = [ os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav") for x in fi.read().split("\n") if len(x) > 0 ] print(f"first validation file: {validation_files[0]}") for i in range(len(a.list_input_unseen_validation_file)): with open(a.list_input_unseen_validation_file[i], "r", encoding="utf-8") as fi: unseen_validation_files = [ os.path.join(a.list_input_unseen_wavs_dir[i], x.split("|")[0] + ".wav") for x in fi.read().split("\n") if len(x) > 0 ] print( f"first unseen {i}th validation fileset: {unseen_validation_files[0]}" ) list_unseen_validation_files.append(unseen_validation_files) return training_files, validation_files, list_unseen_validation_files class MelDataset(torch.utils.data.Dataset): def __init__( self, training_files, hparams, segment_size, n_fft, num_mels, hop_size, win_size, sampling_rate, fmin, fmax, split=True, shuffle=True, n_cache_reuse=1, device=None, fmax_loss=None, fine_tuning=False, base_mels_path=None, is_seen=True, ): self.audio_files = training_files random.seed(1234) if shuffle: random.shuffle(self.audio_files) self.hparams = hparams self.is_seen = is_seen if self.is_seen: self.name = pathlib.Path(self.audio_files[0]).parts[0] else: self.name = "-".join(pathlib.Path(self.audio_files[0]).parts[:2]).strip("/") self.segment_size = segment_size self.sampling_rate = sampling_rate self.split = split self.n_fft = n_fft self.num_mels = num_mels self.hop_size = hop_size self.win_size = win_size self.fmin = fmin self.fmax = fmax self.fmax_loss = fmax_loss self.cached_wav = None self.n_cache_reuse = n_cache_reuse self._cache_ref_count = 0 self.device = device self.fine_tuning = fine_tuning self.base_mels_path = base_mels_path print("[INFO] checking dataset integrity...") for i in tqdm(range(len(self.audio_files))): assert os.path.exists( self.audio_files[i] ), f"{self.audio_files[i]} not found" def __getitem__(self, index): filename = self.audio_files[index] if self._cache_ref_count == 0: audio, sampling_rate = load_wav(filename, self.sampling_rate) audio = audio / MAX_WAV_VALUE if not self.fine_tuning: audio = normalize(audio) * 0.95 self.cached_wav = audio if sampling_rate != self.sampling_rate: raise ValueError( f"{sampling_rate} SR doesn't match target {self.sampling_rate} SR" ) self._cache_ref_count = self.n_cache_reuse else: audio = self.cached_wav self._cache_ref_count -= 1 audio = torch.FloatTensor(audio) audio = audio.unsqueeze(0) if not self.fine_tuning: if self.split: if audio.size(1) >= self.segment_size: max_audio_start = audio.size(1) - self.segment_size audio_start = random.randint(0, max_audio_start) audio = audio[:, audio_start : audio_start + self.segment_size] else: audio = torch.nn.functional.pad( audio, (0, self.segment_size - audio.size(1)), "constant" ) mel = mel_spectrogram( audio, self.n_fft, self.num_mels, self.sampling_rate, self.hop_size, self.win_size, self.fmin, self.fmax, center=False, ) else: # Validation step # Match audio length to self.hop_size * n for evaluation if (audio.size(1) % self.hop_size) != 0: audio = audio[:, : -(audio.size(1) % self.hop_size)] mel = mel_spectrogram( audio, self.n_fft, self.num_mels, self.sampling_rate, self.hop_size, self.win_size, self.fmin, self.fmax, center=False, ) assert ( audio.shape[1] == mel.shape[2] * self.hop_size ), f"audio shape {audio.shape} mel shape {mel.shape}" else: mel = np.load( os.path.join( self.base_mels_path, os.path.splitext(os.path.split(filename)[-1])[0] + ".npy", ) ) mel = torch.from_numpy(mel) if len(mel.shape) < 3: mel = mel.unsqueeze(0) if self.split: frames_per_seg = math.ceil(self.segment_size / self.hop_size) if audio.size(1) >= self.segment_size: mel_start = random.randint(0, mel.size(2) - frames_per_seg - 1) mel = mel[:, :, mel_start : mel_start + frames_per_seg] audio = audio[ :, mel_start * self.hop_size : (mel_start + frames_per_seg) * self.hop_size, ] else: mel = torch.nn.functional.pad( mel, (0, frames_per_seg - mel.size(2)), "constant" ) audio = torch.nn.functional.pad( audio, (0, self.segment_size - audio.size(1)), "constant" ) mel_loss = mel_spectrogram( audio, self.n_fft, self.num_mels, self.sampling_rate, self.hop_size, self.win_size, self.fmin, self.fmax_loss, center=False, ) return (mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze()) def __len__(self): return len(self.audio_files)