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( "#