diff --git a/README.md b/README.md
index 640021b..794a99f 100644
--- a/README.md
+++ b/README.md
@@ -2,7 +2,8 @@
 Singing Voice Conversion via diffusion model
 
 ## updates:
->2022.11.23 修复了一个重大bug，曾导致可能将用于推理的原始gt音频转变采样率为22.05kHz,对于由此造成的影响我们表示十分抱歉，请务必检查自己的测试音频，并使用更新后的代码\
+>2022.11.28 增加了默认打开的no_fs2选项，可优化部分网络，提升训练速度、缩减模型体积，对于未来新训练的模型有效\
+2022.11.23 修复了一个重大bug，曾导致可能将用于推理的原始gt音频转变采样率为22.05kHz,对于由此造成的影响我们表示十分抱歉，请务必检查自己的测试音频，并使用更新后的代码\
 2022.11.22 修复了很多bug，其中有几个影响推理效果重大的bug\
 2022.11.20 增加对推理时多数格式的输入和保存，无需手动借助其他软件转换\
 2022.11.13 修正中断后读取模型的epoch/steps显示问题，添加f0处理的磁盘缓存，添加实时变声推理的支持文件\
diff --git a/infer.py b/infer.py
index 0d4f3b2..b5ceecc 100644
--- a/infer.py
+++ b/infer.py
@@ -42,7 +42,7 @@ def run_clip(svc_model, key, acc, use_pe, use_crepe, thre, use_gt_mel, add_noise
     f0_tst = []
     f0_pred = []
     audio = []
-    epsilon = 0.0002
+    epsilon = 0.00002
     for data in audio_data:
         print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======')
         length = int(np.ceil(len(data) / audio_sr * hparams['audio_sample_rate']))
diff --git a/modules/fastspeech/fs2.py b/modules/fastspeech/fs2.py
index 1dce7bf..e718e00 100644
--- a/modules/fastspeech/fs2.py
+++ b/modules/fastspeech/fs2.py
@@ -23,12 +23,13 @@ def __init__(self, dictionary, out_dims=None):
         super().__init__()
         # self.dictionary = dictionary
         self.padding_idx = 0
-        self.enc_layers = hparams['enc_layers']
-        self.dec_layers = hparams['dec_layers']
+        if hparams['no_fs2'] if 'no_fs2' in hparams.keys() else True:
+            self.enc_layers = hparams['enc_layers']
+            self.dec_layers = hparams['dec_layers']
+            self.encoder = FS_ENCODERS[hparams['encoder_type']](hparams)
+            self.decoder = FS_DECODERS[hparams['decoder_type']](hparams)
         self.hidden_size = hparams['hidden_size']
         # self.encoder_embed_tokens = self.build_embedding(self.dictionary, self.hidden_size)
-        self.encoder = FS_ENCODERS[hparams['encoder_type']](hparams)
-        self.decoder = FS_DECODERS[hparams['decoder_type']](hparams)
         self.out_dims = out_dims
         if out_dims is None:
             self.out_dims = hparams['audio_num_mel_bins']
@@ -94,7 +95,10 @@ def forward(self, hubert, mel2ph=None, spk_embed=None,
                 ref_mels=None, f0=None, uv=None, energy=None, skip_decoder=True,
                 spk_embed_dur_id=None, spk_embed_f0_id=None, infer=False, **kwargs):
         ret = {}
-        encoder_out =self.encoder(hubert)  # [B, T, C]
+        if hparams['no_fs2'] if 'no_fs2' in hparams.keys() else True:
+            encoder_out =self.encoder(hubert)  # [B, T, C]
+        else:
+            encoder_out =hubert
         src_nonpadding = (hubert!=0).any(-1)[:,:,None]
 
         # add ref style embed
@@ -142,10 +146,10 @@ def forward(self, hubert, mel2ph=None, spk_embed=None,
             decoder_inp = decoder_inp + self.add_energy(pitch_inp, energy, ret)
 
         ret['decoder_inp'] = decoder_inp = (decoder_inp + spk_embed) * tgt_nonpadding
-
-        if skip_decoder:
-            return ret
-        ret['mel_out'] = self.run_decoder(decoder_inp, tgt_nonpadding, ret, infer=infer, **kwargs)
+        if hparams['no_fs2'] if 'no_fs2' in hparams.keys() else True:
+            if skip_decoder:
+                return ret
+            ret['mel_out'] = self.run_decoder(decoder_inp, tgt_nonpadding, ret, infer=infer, **kwargs)
 
         return ret
 
diff --git a/training/config.yaml b/training/config.yaml
index 5030a2b..e8e3bfd 100644
--- a/training/config.yaml
+++ b/training/config.yaml
@@ -346,3 +346,4 @@ wav2spec_eps: 1e-6
 weight_decay: 0
 win_size: 512
 work_dir: checkpoints/atri
+no_fs2: true
diff --git a/training/config_nsf.yaml b/training/config_nsf.yaml
index 5fda5a6..c4a2e71 100644
--- a/training/config_nsf.yaml
+++ b/training/config_nsf.yaml
@@ -188,3 +188,4 @@ wav2spec_eps: 1e-6
 weight_decay: 0
 win_size: 2048
 work_dir: checkpoints/nyaru
+no_fs2: true