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import numpy as np
import os
import random
import librosa
import parselmouth
import argparse
from pathlib import Path
from multiprocessing import Process
from tqdm import tqdm
from utils.spectrogram import AWeightingLoudness
from utils.tools import load_wav
np.random.seed(0)
random.seed(0)
def extract_loudness(wav_path: str | Path, ld_path: str | Path=None, frame_period=0.01, factor=0):
"""
Extracts loudness information from an audio waveform.
Args:
wav_path (str or Path): Path to the audio waveform file (must be 24kHz)
ld_path (str or Path, optional): Path to load or save the extracted loudness information as a numpy file.
If specified, the function will first attempt to load the loudness information from this path.
If the file does not exist, the loudness will be calculated and saved to this path.
Defaults to None.
frame_period (float, optional): Time duration in seconds for each frame. Defaults to 0.01.
factor (float, optional): Loudness adjustment factor to fit different persons. Defaults to 0.
Returns:
numpy.ndarray: Extracted loudness information.
"""
if ld_path is not None and os.path.isfile(ld_path):
loudness = np.load(ld_path)
return loudness
else:
# extract loudness using 24kHz audio
wav, fs = load_wav(wav_path, 24000)
loudness = AWeightingLoudness(
x=wav,
sr=fs,
n_fft=2048,
n_shift=int(fs*frame_period),
win_length=2048,
window='hann',
)
loudness = loudness + factor
if ld_path is not None:
os.makedirs(ld_path.parent, exist_ok=True)
np.save(ld_path, loudness)
return loudness
def REAPER_F0(wav_path, sr=24000, frame_period=0.01): # frame_period s
if not os.path.isfile(f'{wav_path}.f0'):
cmd = f'REAPER/build/reaper -i {wav_path} -f {wav_path}.f0 -e {frame_period} -x 1000 -m 65 -a'
os.system(cmd)
f0 = []
with open(f'{wav_path}.f0', 'r') as rf:
for line in rf.readlines()[7:]:
f0.append(float(line.split()[2]))
cmd = f'rm -f {wav_path}.f0'
os.system(cmd)
f0 = np.array(f0)
minus_one_indexes = (f0 == -1)
f0[minus_one_indexes] = 0
return f0
def ParselMouth_F0(wav, sr=24000, frame_period=0.01):
wav = parselmouth.Sound(wav, sampling_frequency=sr)
pitch = wav.to_pitch(time_step=frame_period, pitch_floor=65, pitch_ceiling=1000)
f0 = pitch.selected_array['frequency']
return f0
def PYIN_F0(wav, sr=24000, frame_period=10):
fmin = librosa.note_to_hz('C2') # ~65Hz
fmax = librosa.note_to_hz('C7') # ~2093Hz
# fmax = fs/2
f0, voiced_flag, voiced_prob = librosa.pyin(
wav, fmin=fmin, fmax=fmax, sr=sr, frame_length=int(sr*frame_period/1000*4))
f0 = np.where(np.isnan(f0), 0.0, f0)
return f0
def pad_arrays(arrays: list[np.ndarray], std_len: int):
"""
Pad arrays value to a specified standard length.
Args:
arrays (List[numpy.ndarray]): List of arrays to be padded.
std_len (int): Standard length to which the arrays will be padded.
Returns:
List[numpy.ndarray]: List of padded arrays.
Raises:
ValueError: If the length of any array in the input list is greater than the specified standard length.
"""
padded_arrays = []
for arr in arrays:
cur_len = len(arr)
if cur_len <= std_len:
pad_width = std_len - cur_len
left_pad = pad_width // 2
right_pad = pad_width - left_pad
padded_arr = np.pad(arr, (left_pad, right_pad), 'edge')
padded_arrays.append(padded_arr)
else:
raise ValueError(f'cur_len: {cur_len}, std_len: {std_len}.')
return padded_arrays
def compute_pitch(wav_path: str, pitch_path: str=None, frame_period=0.01):
"""
Computes the pitch information from an audio waveform.
Args:
wav_path (str): Path to the audio waveform file (must be 24kHz).
pitch_path (str, optional): Path to save or load the computed pitch information as a numpy file.
If specified, the function will first attempt to load the pitch information from this path.
If the file does not exist, the pitch will be computed and saved to this path.
Defaults to None.
frame_period (float, optional): Time duration in seconds for each frame. Defaults to 0.01.
Returns:
numpy.ndarray: Computed pitch information.
Notes:
For precise pitch representation, the pitch values are extracted by the median of three methods:
the PYIN, the REAPER, and the Parselmouth.
"""
if pitch_path is not None and os.path.isfile(pitch_path):
pitch = np.load(pitch_path)
return pitch
else:
# extract pitch using 24kHz audio
wav, fs = load_wav(wav_path, 24000)
f0_std_len = wav.shape[0] // int(frame_period*fs) + 1
compute_median = []
# Compute pitch using PYIN algorithm
f0 = PYIN_F0(wav, sr=fs, frame_period=frame_period*1000)
compute_median.append(f0)
# Compute pitch using ParselMouth algorithm
f0 = ParselMouth_F0(wav, sr=fs, frame_period=frame_period)
compute_median.append(f0)
# Compute pitch using REAPER algorithm
f0 = REAPER_F0(wav_path, sr=fs, frame_period=frame_period)
compute_median.append(f0)
# Compute median F0
compute_median = pad_arrays(compute_median, f0_std_len)
compute_median = np.array(compute_median)
median_f0 = np.median(compute_median, axis=0)
if pitch_path is not None:
os.makedirs(pitch_path.parent, exist_ok=True)
np.save(pitch_path, median_f0)
return median_f0
def extract_pitch_ref(wav_path: str, ref_path: str, predefined_factor=0, speech_enroll=False):
"""
Extracts pitch information from an audio waveform and adjusts it based on a reference audio.
Args:
wav_path (str): Path to the audio waveform file.
ref_path (str): Path to the reference audio waveform file.
predefined_factor (float, optional): Predefined factor to adjust the pitch.
If non-zero, this factor will be used instead of computing it from the reference audio. Defaults to 0.
speech_enroll (bool, optional): Flag indicating whether the pitch adjustment is for speech enrollment. Defaults to False.
Returns:
Tuple[numpy.ndarray, float]: Tuple containing the adjusted pitch information (source_f0) and the pitch shift factor (factor).
"""
source_f0 = compute_pitch(wav_path)
nonzero_indices = np.nonzero(source_f0)
source_mean = np.mean(source_f0[nonzero_indices], axis=0)
if predefined_factor != 0.:
print(f'Using predefined factor {predefined_factor}.')
factor = predefined_factor
else:
# Compute mean and std for pitch with the reference audio
ref_wav, fs = load_wav(ref_path)
ref_f0 = ParselMouth_F0(ref_wav, fs)
nonzero_indices = np.nonzero(ref_f0)
ref_mean = np.mean(ref_f0[nonzero_indices], axis=0)
factor = ref_mean / source_mean
if speech_enroll:
factor = factor * 1.2
print(f'pitch shift factor: {factor:.2f}')
# Modify f0 to fit with different persons
source_f0 = source_f0 * factor
return source_f0, factor
def go(files, audio_dir, pitch_dir, ld_dir, rank):
if rank == 0:
pb = tqdm(files)
else:
pb = files
for file in pb:
ld = extract_loudness(file, (ld_dir/file.relative_to(audio_dir)).with_suffix('.npy'))
f0 = compute_pitch(file, (pitch_dir/file.relative_to(audio_dir)).with_suffix('.npy'))
def main(args):
data_root = Path(args.data_root)
pitch_dir = Path(args.pitch_dir) if args.pitch_dir is not None else data_root/'pitch'
ld_dir = Path(args.ld_dir) if args.ld_dir is not None else data_root/'loudness'
n_p = args.n_cpu
files = list(data_root.rglob('*.wav'))
print(f"{len(files)} files to extract")
ps = []
for i in range(n_p):
p = Process(
target=go,
args=(files[i::n_p], data_root, pitch_dir, ld_dir, i)
)
ps.append(p)
p.start()
for i in range(n_p):
ps[i].join()
if __name__ == '__main__':
parser = argparse.ArgumentParser(description="Compute pitch and loudness")
parser.add_argument('--data_root', required=True, type=str)
parser.add_argument('--pitch_dir', type=str)
parser.add_argument('--ld_dir', type=str)
parser.add_argument('--n_cpu', type=int, default=1)
args = parser.parse_args()
main(args)
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