Format code (#275)

Co-authored-by: github-actions[bot] <github-actions[bot]@users.noreply.github.com>

vc_infer_pipeline.py

@@ -2,16 +2,18 @@ import numpy as np, parselmouth, torch, pdb
 from time import time as ttime
 import torch.nn.functional as F
 import scipy.signal as signal
-import pyworld, os, traceback, faiss,librosa
+import pyworld, os, traceback, faiss, librosa
 from scipy import signal
 from functools import lru_cache
 
 bh, ah = signal.butter(N=5, Wn=48, btype="high", fs=16000)
 
-input_audio_path2wav={}
+input_audio_path2wav = {}
+
+
 @lru_cache
-def cache_harvest_f0(input_audio_path,fs,f0max,f0min,frame_period):
-    audio=input_audio_path2wav[input_audio_path]
+def cache_harvest_f0(input_audio_path, fs, f0max, f0min, frame_period):
+    audio = input_audio_path2wav[input_audio_path]
     f0, t = pyworld.harvest(
         audio,
         fs=fs,
@@ -22,18 +24,29 @@ def cache_harvest_f0(input_audio_path,fs,f0max,f0min,frame_period):
     f0 = pyworld.stonemask(audio, f0, t, fs)
     return f0
 
-def change_rms(data1,sr1,data2,sr2,rate):#1是输入音频，2是输出音频,rate是2的占比
+
+def change_rms(data1, sr1, data2, sr2, rate):  # 1是输入音频，2是输出音频,rate是2的占比
     # print(data1.max(),data2.max())
-    rms1 = librosa.feature.rms(y=data1, frame_length=sr1//2*2, hop_length=sr1//2)#每半秒一个点
-    rms2 = librosa.feature.rms(y=data2, frame_length=sr2//2*2, hop_length=sr2//2)
-    rms1=torch.from_numpy(rms1)
-    rms1=F.interpolate(rms1.unsqueeze(0), size=data2.shape[0],mode='linear').squeeze()
-    rms2=torch.from_numpy(rms2)
-    rms2=F.interpolate(rms2.unsqueeze(0), size=data2.shape[0],mode='linear').squeeze()
-    rms2=torch.max(rms2,torch.zeros_like(rms2)+1e-6)
-    data2*=(torch.pow(rms1,torch.tensor(1-rate))*torch.pow(rms2,torch.tensor(rate-1))).numpy()
+    rms1 = librosa.feature.rms(
+        y=data1, frame_length=sr1 // 2 * 2, hop_length=sr1 // 2
+    )  # 每半秒一个点
+    rms2 = librosa.feature.rms(y=data2, frame_length=sr2 // 2 * 2, hop_length=sr2 // 2)
+    rms1 = torch.from_numpy(rms1)
+    rms1 = F.interpolate(
+        rms1.unsqueeze(0), size=data2.shape[0], mode="linear"
+    ).squeeze()
+    rms2 = torch.from_numpy(rms2)
+    rms2 = F.interpolate(
+        rms2.unsqueeze(0), size=data2.shape[0], mode="linear"
+    ).squeeze()
+    rms2 = torch.max(rms2, torch.zeros_like(rms2) + 1e-6)
+    data2 *= (
+        torch.pow(rms1, torch.tensor(1 - rate))
+        * torch.pow(rms2, torch.tensor(rate - 1))
+    ).numpy()
     return data2
 
+
 class VC(object):
     def __init__(self, tgt_sr, config):
         self.x_pad, self.x_query, self.x_center, self.x_max, self.is_half = (
@@ -53,7 +66,16 @@ class VC(object):
         self.t_max = self.sr * self.x_max  # 免查询时长阈值
         self.device = config.device
 
-    def get_f0(self, input_audio_path,x, p_len, f0_up_key, f0_method,filter_radius, inp_f0=None):
+    def get_f0(
+        self,
+        input_audio_path,
+        x,
+        p_len,
+        f0_up_key,
+        f0_method,
+        filter_radius,
+        inp_f0=None,
+    ):
         global input_audio_path2wav
         time_step = self.window / self.sr * 1000
         f0_min = 50
@@ -77,9 +99,9 @@ class VC(object):
                     f0, [[pad_size, p_len - len(f0) - pad_size]], mode="constant"
                 )
         elif f0_method == "harvest":
-            input_audio_path2wav[input_audio_path]=x.astype(np.double)
-            f0=cache_harvest_f0(input_audio_path,self.sr,f0_max,f0_min,10)
-            if(filter_radius>2):
+            input_audio_path2wav[input_audio_path] = x.astype(np.double)
+            f0 = cache_harvest_f0(input_audio_path, self.sr, f0_max, f0_min, 10)
+            if filter_radius > 2:
                 f0 = signal.medfilt(f0, 3)
         f0 *= pow(2, f0_up_key / 12)
         # with open("test.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
@@ -134,12 +156,12 @@ class VC(object):
         inputs = {
             "source": feats.to(self.device),
             "padding_mask": padding_mask,
-            "output_layer": 9if version=="v1"else 12,
+            "output_layer": 9 if version == "v1" else 12,
         }
         t0 = ttime()
         with torch.no_grad():
             logits = model.extract_features(**inputs)
-            feats = model.final_proj(logits[0])if version=="v1"else logits[0]
+            feats = model.final_proj(logits[0]) if version == "v1" else logits[0]
 
         if (
             isinstance(index, type(None)) == False
@@ -184,10 +206,7 @@ class VC(object):
                 )
             else:
                 audio1 = (
-                    (net_g.infer(feats, p_len, sid)[0][0, 0])
-                    .data.cpu()
-                    .float()
-                    .numpy()
+                    (net_g.infer(feats, p_len, sid)[0][0, 0]).data.cpu().float().numpy()
                 )
         del feats, p_len, padding_mask
         if torch.cuda.is_available():
@@ -270,7 +289,15 @@ class VC(object):
         sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
         pitch, pitchf = None, None
         if if_f0 == 1:
-            pitch, pitchf = self.get_f0(input_audio_path,audio_pad, p_len, f0_up_key, f0_method,filter_radius, inp_f0)
+            pitch, pitchf = self.get_f0(
+                input_audio_path,
+                audio_pad,
+                p_len,
+                f0_up_key,
+                f0_method,
+                filter_radius,
+                inp_f0,
+            )
             pitch = pitch[:p_len]
             pitchf = pitchf[:p_len]
             if self.device == "mps":
@@ -347,16 +374,17 @@ class VC(object):
                 )[self.t_pad_tgt : -self.t_pad_tgt]
             )
         audio_opt = np.concatenate(audio_opt)
-        if(rms_mix_rate!=1):
-            audio_opt=change_rms(audio,16000,audio_opt,tgt_sr,rms_mix_rate)
-        if(resample_sr>=16000 and tgt_sr!=resample_sr):
+        if rms_mix_rate != 1:
+            audio_opt = change_rms(audio, 16000, audio_opt, tgt_sr, rms_mix_rate)
+        if resample_sr >= 16000 and tgt_sr != resample_sr:
             audio_opt = librosa.resample(
                 audio_opt, orig_sr=tgt_sr, target_sr=resample_sr
             )
-        audio_max=np.abs(audio_opt).max()/0.99
-        max_int16=32768
-        if(audio_max>1):max_int16/=audio_max
-        audio_opt=(audio_opt * max_int16).astype(np.int16)
+        audio_max = np.abs(audio_opt).max() / 0.99
+        max_int16 = 32768
+        if audio_max > 1:
+            max_int16 /= audio_max
+        audio_opt = (audio_opt * max_int16).astype(np.int16)
         del pitch, pitchf, sid
         if torch.cuda.is_available():
             torch.cuda.empty_cache()