Jesus Lopez
feat: applio
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import os
import sys
import time
from scipy import signal
from scipy.io import wavfile
import numpy as np
import concurrent.futures
from tqdm import tqdm
import json
from distutils.util import strtobool
import librosa
import multiprocessing
import noisereduce as nr
now_directory = os.getcwd()
sys.path.append(now_directory)
from rvc.lib.utils import load_audio
from rvc.train.preprocess.slicer import Slicer
# Remove colab logs
import logging
logging.getLogger("numba.core.byteflow").setLevel(logging.WARNING)
logging.getLogger("numba.core.ssa").setLevel(logging.WARNING)
logging.getLogger("numba.core.interpreter").setLevel(logging.WARNING)
# Constants
OVERLAP = 0.3
MAX_AMPLITUDE = 0.9
ALPHA = 0.75
HIGH_PASS_CUTOFF = 48
SAMPLE_RATE_16K = 16000
class PreProcess:
def __init__(self, sr: int, exp_dir: str, per: float):
self.slicer = Slicer(
sr=sr,
threshold=-42,
min_length=1500,
min_interval=400,
hop_size=15,
max_sil_kept=500,
)
self.sr = sr
self.b_high, self.a_high = signal.butter(
N=5, Wn=HIGH_PASS_CUTOFF, btype="high", fs=self.sr
)
self.per = per
self.exp_dir = exp_dir
self.device = "cpu"
self.gt_wavs_dir = os.path.join(exp_dir, "sliced_audios")
self.wavs16k_dir = os.path.join(exp_dir, "sliced_audios_16k")
os.makedirs(self.gt_wavs_dir, exist_ok=True)
os.makedirs(self.wavs16k_dir, exist_ok=True)
def _normalize_audio(self, audio: np.ndarray):
tmp_max = np.abs(audio).max()
if tmp_max > 2.5:
return None
return (audio / tmp_max * (MAX_AMPLITUDE * ALPHA)) + (1 - ALPHA) * audio
def process_audio_segment(
self,
normalized_audio: np.ndarray,
sid: int,
idx0: int,
idx1: int,
):
if normalized_audio is None:
print(f"{sid}-{idx0}-{idx1}-filtered")
return
wavfile.write(
os.path.join(self.gt_wavs_dir, f"{sid}_{idx0}_{idx1}.wav"),
self.sr,
normalized_audio.astype(np.float32),
)
audio_16k = librosa.resample(
normalized_audio, orig_sr=self.sr, target_sr=SAMPLE_RATE_16K
)
wavfile.write(
os.path.join(self.wavs16k_dir, f"{sid}_{idx0}_{idx1}.wav"),
SAMPLE_RATE_16K,
audio_16k.astype(np.float32),
)
def process_audio(
self,
path: str,
idx0: int,
sid: int,
cut_preprocess: bool,
process_effects: bool,
noise_reduction: bool,
reduction_strength: float,
):
audio_length = 0
try:
audio = load_audio(path, self.sr)
audio_length = librosa.get_duration(y=audio, sr=self.sr)
if process_effects:
audio = signal.lfilter(self.b_high, self.a_high, audio)
audio = self._normalize_audio(audio)
if noise_reduction:
audio = nr.reduce_noise(
y=audio, sr=self.sr, prop_decrease=reduction_strength
)
idx1 = 0
if cut_preprocess:
for audio_segment in self.slicer.slice(audio):
i = 0
while True:
start = int(self.sr * (self.per - OVERLAP) * i)
i += 1
if len(audio_segment[start:]) > (self.per + OVERLAP) * self.sr:
tmp_audio = audio_segment[
start : start + int(self.per * self.sr)
]
self.process_audio_segment(
tmp_audio,
sid,
idx0,
idx1,
)
idx1 += 1
else:
tmp_audio = audio_segment[start:]
self.process_audio_segment(
tmp_audio,
sid,
idx0,
idx1,
)
idx1 += 1
break
else:
self.process_audio_segment(
audio,
sid,
idx0,
idx1,
)
except Exception as error:
print(f"Error processing audio: {error}")
return audio_length
def format_duration(seconds):
hours = int(seconds // 3600)
minutes = int((seconds % 3600) // 60)
seconds = int(seconds % 60)
return f"{hours:02}:{minutes:02}:{seconds:02}"
def save_dataset_duration(file_path, dataset_duration):
try:
with open(file_path, "r") as f:
data = json.load(f)
except FileNotFoundError:
data = {}
formatted_duration = format_duration(dataset_duration)
new_data = {
"total_dataset_duration": formatted_duration,
"total_seconds": dataset_duration,
}
data.update(new_data)
with open(file_path, "w") as f:
json.dump(data, f, indent=4)
def process_audio_wrapper(args):
pp, file, cut_preprocess, process_effects, noise_reduction, reduction_strength = (
args
)
file_path, idx0, sid = file
return pp.process_audio(
file_path,
idx0,
sid,
cut_preprocess,
process_effects,
noise_reduction,
reduction_strength,
)
def preprocess_training_set(
input_root: str,
sr: int,
num_processes: int,
exp_dir: str,
per: float,
cut_preprocess: bool,
process_effects: bool,
noise_reduction: bool,
reduction_strength: float,
):
start_time = time.time()
pp = PreProcess(sr, exp_dir, per)
print(f"Starting preprocess with {num_processes} processes...")
files = []
idx = 0
for root, _, filenames in os.walk(input_root):
try:
sid = 0 if root == input_root else int(os.path.basename(root))
for f in filenames:
if f.lower().endswith((".wav", ".mp3", ".flac", ".ogg")):
files.append((os.path.join(root, f), idx, sid))
idx += 1
except ValueError:
print(
f'Speaker ID folder is expected to be integer, got "{os.path.basename(root)}" instead.'
)
# print(f"Number of files: {len(files)}")
audio_length = []
with tqdm(total=len(files)) as pbar:
with concurrent.futures.ProcessPoolExecutor(
max_workers=num_processes
) as executor:
futures = [
executor.submit(
process_audio_wrapper,
(
pp,
file,
cut_preprocess,
process_effects,
noise_reduction,
reduction_strength,
),
)
for file in files
]
for future in concurrent.futures.as_completed(futures):
audio_length.append(future.result())
pbar.update(1)
audio_length = sum(audio_length)
save_dataset_duration(
os.path.join(exp_dir, "model_info.json"), dataset_duration=audio_length
)
elapsed_time = time.time() - start_time
print(
f"Preprocess completed in {elapsed_time:.2f} seconds on {format_duration(audio_length)} seconds of audio."
)
if __name__ == "__main__":
experiment_directory = str(sys.argv[1])
input_root = str(sys.argv[2])
sample_rate = int(sys.argv[3])
percentage = float(sys.argv[4])
num_processes = sys.argv[5]
if num_processes.lower() == "none":
num_processes = multiprocessing.cpu_count()
else:
num_processes = int(num_processes)
cut_preprocess = strtobool(sys.argv[6])
process_effects = strtobool(sys.argv[7])
noise_reduction = strtobool(sys.argv[8])
reduction_strength = float(sys.argv[9])
preprocess_training_set(
input_root,
sample_rate,
num_processes,
experiment_directory,
percentage,
cut_preprocess,
process_effects,
noise_reduction,
reduction_strength,
)