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import os
import io
import gradio as gr
import librosa
import numpy as np
import logging
import soundfile
import torchaudio
import asyncio
import argparse
import subprocess
import gradio.processing_utils as gr_processing_utils
logging.getLogger('numba').setLevel(logging.WARNING)
logging.getLogger('markdown_it').setLevel(logging.WARNING)
logging.getLogger('urllib3').setLevel(logging.WARNING)
logging.getLogger('matplotlib').setLevel(logging.WARNING)
limitation = os.getenv("SYSTEM") == "spaces" # limit audio length in huggingface spaces
def unused_vc_fn(input_audio, vc_transform, voice):
if input_audio is None:
return "You need to upload an audio", None
sampling_rate, audio = input_audio
duration = audio.shape[0] / sampling_rate
if duration > 20 and limitation:
return "Please upload an audio file that is less than 20 seconds. If you need to generate a longer audio file, please use Colab.", None
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != 16000:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
raw_path = io.BytesIO()
soundfile.write(raw_path, audio, 16000, format="wav")
raw_path.seek(0)
out_audio, out_sr = model.infer(sid, vc_transform, raw_path,
auto_predict_f0=True,
)
return "Success", (44100, out_audio.cpu().numpy())
def run_inference(input_audio, speaker):
if input_audio is None:
return "You need to upload an audio", None
sampling_rate, audio = input_audio
duration = audio.shape[0] / sampling_rate
if duration > 20 and limitation:
return "Please upload an audio file that is less than 20 seconds. If you need to generate a longer audio file, please use Colab.", None
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != 16000:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
#TODO edit from GUI
cluster_ratio = 1
noise_scale = 2
is_pitch_prediction_enabled = True
f0_method = "dio"
transpose = 0
model_path = f"./models/{speaker}/{speaker}.pth"
config_path = f"./models/{speaker}/config.json"
cluster_path = ""
raw_path = 'tmp.wav'
soundfile.write(raw_path, audio, 16000, format="wav")
inference_cmd = f"svc infer {raw_path} -m {model_path} -c {config_path} {f'-k {cluster_path} -r {cluster_ratio}' if cluster_path != '' and cluster_ratio > 0 else ''} -t {transpose} --f0-method {f0_method} -n {noise_scale} -o out.wav {'' if is_pitch_prediction_enabled else '--no-auto-predict-f0'}"
print(inference_cmd)
result = subprocess.run(
inference_cmd.split(),
stdout=subprocess.PIPE,
stderr=subprocess.STDOUT,
text=True
)
audio, sr = torchaudio.load('out.wav')
out_audio = audio.cpu().numpy()[0]
print(out_audio)
return 'out.wav' # (sr, out_audio)
if __name__ == '__main__':
parser = argparse.ArgumentParser()
parser.add_argument('--device', type=str, default='cpu')
parser.add_argument('--api', action="store_true", default=False)
parser.add_argument("--share", action="store_true", default=False, help="share gradio app")
args = parser.parse_args()
speakers = ["chapaev", "petka", "anka", "narrator", "floppa"]
models = []
voices = []
# !svc infer {NAME}.wav -c config.json -m G_riri_220.pth
# display(Audio(f"{NAME}.out.wav", autoplay=True))
with gr.Blocks() as app:
gr.Markdown(
"# <center> Sovits Chapay\n"
)
with gr.Row():
with gr.Column():
vc_input = gr.Audio(label="Input audio"+' (less than 20 seconds)' if limitation else '')
speaker = gr.Dropdown(label="Speaker", choices=speakers, visible=True)
vc_submit = gr.Button("Generate", variant="primary")
with gr.Column():
vc_output = gr.Audio(label="Output Audio")
vc_submit.click(run_inference, [vc_input, speaker], [vc_output])
app.queue(concurrency_count=1, api_open=True).launch(show_api=True, show_error=True)
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