diff --git a/inference/infer_tool.py b/inference/infer_tool.py
index 6440fb0..dd1799a 100644
--- a/inference/infer_tool.py
+++ b/inference/infer_tool.py
@@ -190,38 +190,59 @@ class Svc(object):
# 清理显存
torch.cuda.empty_cache()
- def slice_inference(self,raw_audio_path, spk, tran, slice_db,cluster_infer_ratio, auto_predict_f0,noice_scale, pad_seconds=0.5):
+ def slice_inference(self,raw_audio_path, spk, tran, slice_db,cluster_infer_ratio, auto_predict_f0,noice_scale, pad_seconds=0.5, clip_seconds=0,lg_num=0,lgr_num =0.75):
wav_path = raw_audio_path
chunks = slicer.cut(wav_path, db_thresh=slice_db)
audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks)
-
+ per_size = int(clip_seconds*audio_sr)
+ lg_size = int(lg_num*audio_sr)
+ lg_size_r = int(lg_size*lgr_num)
+ lg_size_c_l = (lg_size-lg_size_r)//2
+ lg_size_c_r = lg_size-lg_size_r-lg_size_c_l
+ lg = np.linspace(0,1,lg_size_r) if lg_size!=0 else 0
+
audio = []
for (slice_tag, data) in audio_data:
print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======')
# padd
- pad_len = int(audio_sr * pad_seconds)
- data = np.concatenate([np.zeros([pad_len]), data, np.zeros([pad_len])])
length = int(np.ceil(len(data) / audio_sr * self.target_sample))
- raw_path = io.BytesIO()
- soundfile.write(raw_path, data, audio_sr, format="wav")
- raw_path.seek(0)
if slice_tag:
print('jump empty segment')
_audio = np.zeros(length)
+ audio.extend(list(pad_array(_audio, length)))
+ continue
+ if per_size != 0:
+ datas = split_list_by_n(data, per_size,lg_size)
else:
+ datas = [data]
+ for k,dat in enumerate(datas):
+ per_length = int(np.ceil(len(dat) / audio_sr * self.target_sample)) if clip_seconds!=0 else length
+ if clip_seconds!=0: print(f'###=====segment clip start, {round(len(dat) / audio_sr, 3)}s======')
+ # padd
+ pad_len = int(audio_sr * pad_seconds)
+ dat = np.concatenate([np.zeros([pad_len]), dat, np.zeros([pad_len])])
+ raw_path = io.BytesIO()
+ soundfile.write(raw_path, dat, audio_sr, format="wav")
+ raw_path.seek(0)
out_audio, out_sr = self.infer(spk, tran, raw_path,
cluster_infer_ratio=cluster_infer_ratio,
auto_predict_f0=auto_predict_f0,
noice_scale=noice_scale
)
_audio = out_audio.cpu().numpy()
-
- pad_len = int(self.target_sample * pad_seconds)
- _audio = _audio[pad_len:-pad_len]
- audio.extend(list(_audio))
+ pad_len = int(self.target_sample * pad_seconds)
+ _audio = _audio[pad_len:-pad_len]
+ _audio = pad_array(_audio, per_length)
+ if lg_size!=0 and k!=0:
+ lg1 = audio[-(lg_size_r+lg_size_c_r):-lg_size_c_r] if lgr_num != 1 else audio[-lg_size:]
+ lg2 = _audio[lg_size_c_l:lg_size_c_l+lg_size_r] if lgr_num != 1 else _audio[0:lg_size]
+ lg_pre = lg1*(1-lg)+lg2*lg
+ audio = audio[0:-(lg_size_r+lg_size_c_r)] if lgr_num != 1 else audio[0:-lg_size]
+ audio.extend(lg_pre)
+ _audio = _audio[lg_size_c_l+lg_size_r:] if lgr_num != 1 else _audio[lg_size:]
+ audio.extend(list(_audio))
return np.array(audio)
-
class RealTimeVC:
def __init__(self):
self.last_chunk = None
diff --git a/webUI.py b/webUI.py
new file mode 100644
index 0000000..1795982
--- /dev/null
+++ b/webUI.py
@@ -0,0 +1,94 @@
+import io
+import os
+
+# os.system("wget -P cvec/ https://huggingface.co/spaces/innnky/nanami/resolve/main/checkpoint_best_legacy_500.pt")
+import gradio as gr
+import librosa
+import numpy as np
+import soundfile
+from inference.infer_tool import Svc
+import logging
+
+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)
+logging.getLogger('multipart').setLevel(logging.WARNING)
+
+model = None
+spk = None
+
+def vc_fn(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num):
+ global model
+ try:
+ if input_audio is None:
+ return "You need to upload an audio", None
+ if model is None:
+ return "You need to upload an model", None
+ sampling_rate, audio = input_audio
+ # print(audio.shape,sampling_rate)
+ audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
+ if len(audio.shape) > 1:
+ audio = librosa.to_mono(audio.transpose(1, 0))
+ temp_path = "temp.wav"
+ soundfile.write(temp_path, audio, model.target_sample, format="wav")
+ _audio = model.slice_inference(temp_path, sid, vc_transform, slice_db, cluster_ratio, auto_f0, noise_scale,pad_seconds,cl_num,lg_num,lgr_num)
+ os.remove(temp_path)
+ return "Success", (model.target_sample, _audio)
+ except Exception as e:
+ return "异常信息:"+str(e)+"\n请排障后重试",None
+
+app = gr.Blocks()
+with app:
+ with gr.Tabs():
+ with gr.TabItem("Sovits4.0"):
+ gr.Markdown(value="""
+ Sovits4.0 WebUI
+ """)
+
+ gr.Markdown(value="""
+ 下面是模型文件选择：
+ """)
+ model_path = gr.File(label="模型文件")
+ gr.Markdown(value="""
+ 下面是配置文件选择：
+ """)
+ config_path = gr.File(label="配置文件")
+ gr.Markdown(value="""
+ 下面是聚类模型文件选择，没有可以不填：
+ """)
+ cluster_model_path = gr.File(label="聚类模型文件")
+ device = gr.Dropdown(label="推理设备，留白则为自动选择cpu和gpu",choices=[None,"gpu","cpu"],value=None)
+ gr.Markdown(value="""
+ 全部上传完毕后(全部文件模块显示download),点击模型解析进行解析：
+ """)
+ model_analysis_button = gr.Button(value="模型解析")
+ sid = gr.Dropdown(label="音色（说话人）")
+ sid_output = gr.Textbox(label="Output Message")
+ vc_input3 = gr.Audio(label="上传音频")
+ vc_transform = gr.Number(label="变调（整数，可以正负，半音数量，升高八度就是12）", value=0)
+ cluster_ratio = gr.Number(label="聚类模型混合比例，0-1之间，默认为0不启用聚类，能提升音色相似度，但会导致咬字下降（如果使用建议0.5左右）", value=0)
+ auto_f0 = gr.Checkbox(label="自动f0预测，配合聚类模型f0预测效果更好,会导致变调功能失效（仅限转换语音，歌声不要勾选此项会究极跑调）", value=False)
+ slice_db = gr.Number(label="切片阈值", value=-40)
+ noise_scale = gr.Number(label="noise_scale 建议不要动，会影响音质，玄学参数", value=0.4)
+ cl_num = gr.Number(label="音频自动切片，0为不切片，单位为秒/s", value=0)
+ pad_seconds = gr.Number(label="推理音频pad秒数，由于未知原因开头结尾会有异响，pad一小段静音段后就不会出现", value=0.5)
+ lg_num = gr.Number(label="两端音频切片的交叉淡入长度，如果自动切片后出现人声不连贯可调整该数值，如果连贯建议采用默认值0，注意，该设置会影响推理速度，单位为秒/s", value=0)
+ lgr_num = gr.Number(label="自动音频切片后，需要舍弃每段切片的头尾。该参数设置交叉长度保留的比例，范围0-1,左开右闭", value=0.75,interactive=True)
+ vc_submit = gr.Button("转换", variant="primary")
+ vc_output1 = gr.Textbox(label="Output Message")
+ vc_output2 = gr.Audio(label="Output Audio")
+ def modelAnalysis(model_path,config_path,cluster_model_path,device):
+ try:
+ global model
+ model = Svc(model_path.name, config_path.name,device=device if device!="" else None,cluster_model_path= cluster_model_path.name if cluster_model_path!=None else "")
+ spks = list(model.spk2id.keys())
+ return sid.update(choices = spks,value=spks[0]),"ok"
+ except Exception as e:
+ return "","异常信息:"+str(e)+"\n请排障后重试"
+ vc_submit.click(vc_fn, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num], [vc_output1, vc_output2])
+ model_analysis_button.click(modelAnalysis,[model_path,config_path,cluster_model_path,device],[sid,sid_output])
+ app.launch()
+
+
+