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import io
import os
os.system("chmod 777 ffmpeg")
import torch
import gradio as gr
import librosa
import numpy as np
import soundfile
import logging
from fairseq import checkpoint_utils
from my_utils import load_audio
from vc_infer_pipeline import VC
import traceback
from config import Config
from infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from i18n import I18nAuto
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)
i18n = I18nAuto()
i18n.print()
config = Config()
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
hubert_model = hubert_model.float()
hubert_model.eval()
global n_spk, tgt_sr, net_g, vc, cpt, version
person = "weights/simple-guitar-crepe-guolv_e1000.pth"
print("loading %s" % person)
cpt = torch.load(person, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=False)
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False))
net_g.eval().to(config.device)
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
version="v2"
default_audio=load_audio("logs/mute/1_16k_wavs/mute.wav",16000)
def vc_single(
# sid=0,
input_audio_path,#待选取
f0_up_key,#待选取
f0_method,
file_index="logs/added_IVF2225_Flat_nprobe_1_simple-guitar-crepe-guolv_v2.index",#写死
index_rate=1,#写死1
filter_radius=3,#不需要,随便写,3
resample_sr=0,#写死0不需要
rms_mix_rate=1,#写死1不需要
protect=0.5,#写死0.5不需要
):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio_path is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
try:
audio = input_audio_path[1] / 32768.0
if len(audio.shape) == 2:
audio = np.mean(audio, -1)
audio = librosa.resample(audio, orig_sr=input_audio_path[0], target_sr=16000)
audio_max = np.abs(audio).max() / 0.95
if audio_max > 1:
audio /= audio_max
times = [0, 0, 0]
audio_opt = vc.pipeline(
model=hubert_model,
net_g=net_g,
sid=0,
audio=audio,
input_audio_path="123",
times=times,
f0_up_key=f0_up_key,
f0_method=f0_method,
file_index=file_index,
index_rate=index_rate,
if_f0=1,
filter_radius=filter_radius,
tgt_sr=tgt_sr,
resample_sr=resample_sr,
rms_mix_rate=rms_mix_rate,
version="v2",
protect=protect,
f0_file=None,
)
if resample_sr >= 16000 and tgt_sr != resample_sr:
tgt_sr = resample_sr
index_info = (
"Using index:%s." % file_index
if os.path.exists(file_index)
else "Index not used."
)
return "Success.\n %s\nTime:\n npy:%ss, f0:%ss, infer:%ss" % (
index_info,
times[0],
times[1],
times[2],
), (tgt_sr, audio_opt)
except:
info = traceback.format_exc()
print(info)
return "报错了!信息如下:%s"%info, (16000, default_audio)
app = gr.Blocks()
with app:
with gr.Tabs():
with gr.TabItem("人声转吉他极简在线demo"):
gr.Markdown(
value="""
变调越高吉他音越细,越低越沉闷
"""
)
vc_input = gr.Audio(label="上传音频")
with gr.Column():
with gr.Row():
vc_transform = gr.Slider(
minimum=-12,
maximum=12,
label="变调(半音数量,升八度12降八度-12)",
value=0,
step=1,
interactive=True,
)
f0method = gr.Radio(
label=i18n(
"选择音高提取算法:语音推荐dio歌声推荐pm"
),
choices=["pm", "dio"],
value="dio",
interactive=True,
)
with gr.Row():
but = gr.Button(i18n("转换"), variant="primary")
vc_output1 = gr.Textbox(label=i18n("输出信息"))
vc_output2 = gr.Audio(label=i18n("输出音频(右下角三个点,点了可以下载)"))
but.click(
vc_single,
[
vc_input,
vc_transform,
f0method
],
[vc_output1, vc_output2],
)
app.launch(server_name="0.0.0.0",quiet=True)
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