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import torch
import numpy as np
from librosa.filters import mel as librosa_mel_fn
from scipy.io.wavfile import read
MAX_WAV_VALUE = 32768.0
def load_wav(full_path):
sampling_rate, data = read(full_path)
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):
output = dynamic_range_compression_torch(magnitudes)
return output
def spectral_de_normalize_torch(magnitudes):
output = dynamic_range_decompression_torch(magnitudes)
return output
mel_basis = {}
hann_window = {}
def mel_spectrogram(y, n_fft=1920, num_mels=80, sampling_rate=24000, hop_size=480,
win_size=1920, fmin=0, fmax=8000, center=False):
global mel_basis, hann_window
if f"{str(fmax)}_{str(y.device)}" not in mel_basis:
mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
mel_basis[str(fmax) + "_" + str(y.device)] = torch.from_numpy(mel).float().to(y.device)
hann_window[str(y.device)] = torch.hann_window(win_size).to(y.device)
y = torch.nn.functional.pad(
y.unsqueeze(1), (int((n_fft - hop_size) / 2), int((n_fft - hop_size) / 2)), mode="reflect"
)
y = y.squeeze(1)
spec = torch.view_as_real(
torch.stft(
y,
n_fft,
hop_length=hop_size,
win_length=win_size,
window=hann_window[str(y.device)],
center=center,
pad_mode="reflect",
normalized=False,
onesided=True,
return_complex=True,
)
)
spec = torch.sqrt(spec.pow(2).sum(-1) + (1e-9))
spec = torch.matmul(mel_basis[str(fmax) + "_" + str(y.device)], spec)
spec = spectral_normalize_torch(spec)
return spec
def audio_volume_normalize(audio: torch.Tensor, coeff=0.1):
"""
Normalize the volume of an audio signal.
Parameters:
audio (torch tensor): Input audio signal array.
coeff (float): Target coefficient for normalization, default is 0.1.
Returns:
torch tensor: The volume-normalized audio signal.
"""
device = audio.device
audio = audio.cpu().numpy()
temp = np.sort(np.abs(audio))
if temp[-1] < 0.1:
scaling_factor = max(
temp[-1], 1e-3
)
audio = audio / scaling_factor * 0.1
temp = temp[temp > 0.01]
L = temp.shape[0]
if L <= 10:
return audio
volume = np.mean(temp[int(0.9 * L) : int(0.99 * L)])
audio = audio * np.clip(coeff / volume, a_min=0.1, a_max=10)
max_value = np.max(np.abs(audio))
if max_value > 1:
audio = audio / max_value
audio = torch.from_numpy(audio).to(device)
return audio