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import os |
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from functools import lru_cache |
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from typing import Union |
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import ffmpeg |
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import numpy as np |
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import torch |
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import torch.nn.functional as F |
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from .utils import exact_div |
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from librosa.filters import mel as librosa_mel_fn |
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SAMPLE_RATE = 16000 |
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N_FFT = 400 |
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N_MELS = 80 |
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HOP_LENGTH = 160 |
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CHUNK_LENGTH = 30 |
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N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE |
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N_FRAMES = exact_div(N_SAMPLES, HOP_LENGTH) |
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def load_audio(file: str, sr: int = SAMPLE_RATE): |
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""" |
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Open an audio file and read as mono waveform, resampling as necessary |
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Parameters |
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---------- |
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file: str |
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The audio file to open |
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sr: int |
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The sample rate to resample the audio if necessary |
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Returns |
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------- |
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A NumPy array containing the audio waveform, in float32 dtype. |
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""" |
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try: |
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out, _ = ( |
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ffmpeg.input(file, threads=0) |
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.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr) |
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.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True) |
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) |
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except ffmpeg.Error as e: |
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e |
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return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0 |
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def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1): |
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""" |
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Pad or trim the audio array to N_SAMPLES, as expected by the encoder. |
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""" |
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if torch.is_tensor(array): |
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if array.shape[axis] > length: |
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array = array.index_select(dim=axis, index=torch.arange(length, device=array.device)) |
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if array.shape[axis] < length: |
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pad_widths = [(0, 0)] * array.ndim |
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pad_widths[axis] = (0, length - array.shape[axis]) |
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array = F.pad(array, [pad for sizes in pad_widths[::-1] for pad in sizes]) |
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else: |
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if array.shape[axis] > length: |
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array = array.take(indices=range(length), axis=axis) |
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if array.shape[axis] < length: |
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pad_widths = [(0, 0)] * array.ndim |
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pad_widths[axis] = (0, length - array.shape[axis]) |
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array = np.pad(array, pad_widths) |
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return array |
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@lru_cache(maxsize=None) |
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def mel_filters(device, n_mels: int = N_MELS) -> torch.Tensor: |
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""" |
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load the mel filterbank matrix for projecting STFT into a Mel spectrogram. |
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Allows decoupling librosa dependency; saved using: |
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np.savez_compressed( |
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"mel_filters.npz", |
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mel_80=librosa.filters.mel(sr=16000, n_fft=400, n_mels=80), |
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) |
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""" |
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assert n_mels == 80, f"Unsupported n_mels: {n_mels}" |
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return torch.from_numpy(librosa_mel_fn(sr=SAMPLE_RATE,n_fft=N_FFT,n_mels=n_mels)).to(device) |
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def log_mel_spectrogram(audio: Union[str, np.ndarray, torch.Tensor], n_mels: int = N_MELS): |
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""" |
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Compute the log-Mel spectrogram of |
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Parameters |
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---------- |
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audio: Union[str, np.ndarray, torch.Tensor], shape = (*) |
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The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz |
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n_mels: int |
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The number of Mel-frequency filters, only 80 is supported |
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Returns |
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------- |
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torch.Tensor, shape = (80, n_frames) |
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A Tensor that contains the Mel spectrogram |
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""" |
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if not torch.is_tensor(audio): |
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if isinstance(audio, str): |
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audio = load_audio(audio) |
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audio = torch.from_numpy(audio) |
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window = torch.hann_window(N_FFT).to(audio.device) |
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stft = torch.stft(audio, N_FFT, HOP_LENGTH, window=window, return_complex=True) |
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magnitudes = stft[..., :-1].abs() ** 2 |
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filters = mel_filters(audio.device, n_mels) |
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mel_spec = filters @ magnitudes |
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log_spec = torch.clamp(mel_spec, min=1e-10).log10() |
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log_spec = torch.maximum(log_spec, log_spec.max() - 8.0) |
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log_spec = (log_spec + 4.0) / 4.0 |
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return log_spec |
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