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import argparse
import os
import sys
from pydub import AudioSegment
import matplotlib.pyplot as plt
import numpy as np
now_dir = os.getcwd()
sys.path.append(now_dir)
from dotenv import load_dotenv
from scipy.io import wavfile
from configs import Config
from infer.modules.vc import VC
####
# USAGE
#
# In your Terminal or CMD or whatever
def arg_parse() -> tuple:
parser = argparse.ArgumentParser()
parser.add_argument("--f0up_key", type=int, default=0)
parser.add_argument("--input_path", type=str, help="input path")
parser.add_argument("--index_path", type=str, help="index path")
parser.add_argument("--f0method", type=str, default="harvest", help="harvest or pm")
parser.add_argument("--opt_path", type=str, help="opt path")
parser.add_argument("--model_name", type=str, help="store in assets/weight_root")
parser.add_argument("--index_rate", type=float, default=0.66, help="index rate")
parser.add_argument("--device", type=str, help="device")
parser.add_argument("--is_half", type=bool, help="use half -> True")
parser.add_argument("--filter_radius", type=int, default=3, help="filter radius")
parser.add_argument("--resample_sr", type=int, default=0, help="resample sr")
parser.add_argument("--rms_mix_rate", type=float, default=1, help="rms mix rate")
parser.add_argument("--protect", type=float, default=0.33, help="protect")
args = parser.parse_args()
sys.argv = sys.argv[:1]
return args
def main():
load_dotenv()
args = arg_parse()
config = Config()
config.device = args.device if args.device else config.device
config.is_half = args.is_half if args.is_half else config.is_half
vc = VC(config)
vc.get_vc(args.model_name)
_, wav_opt = vc.vc_single(
0,
args.input_path,
args.f0up_key,
None,
args.f0method,
args.index_path,
None,
args.index_rate,
args.filter_radius,
args.resample_sr,
args.rms_mix_rate,
args.protect,
)
wavfile.write(args.opt_path, wav_opt[0], wav_opt[1])
# Load the audio file
audio = AudioSegment.from_file(wav_opt) # Replace with your audio file path
# Display basic information about the audio file
print(f"Channels: {audio.channels}")
print(f"Sample Width: {audio.sample_width} bytes")
print(f"Frame Rate (Sample Rate): {audio.frame_rate} Hz")
print(f"Frame Width: {audio.frame_width} bytes")
print(f"Length: {len(audio)} ms")
# Convert the audio data to a numpy array for visualization
samples = np.array(audio.get_array_of_samples())
# If the audio has more than one channel, split the samples into multiple arrays
if audio.channels == 2:
samples = samples.reshape((-1, 2))
# Plot the waveform
plt.figure(figsize=(15, 5))
plt.plot(samples[:1000]) # Plotting first 1000 samples for clarity
plt.title("Waveform of the Audio File")
plt.xlabel("Sample")
plt.ylabel("Amplitude")
plt.show()
if __name__ == "__main__":
main()
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