From bfa45fb0e674537f9d5833f9490c4a94c9c39c07 Mon Sep 17 00:00:00 2001
From: kijai <40791699+kijai@users.noreply.github.com>
Date: Thu, 4 Sep 2025 20:08:26 +0300
Subject: [PATCH] Do interpolation after

---
 comfy/audio_encoders/audio_encoders.py |  9 +++------
 comfy/audio_encoders/wav2vec2.py       | 13 +------------
 2 files changed, 4 insertions(+), 18 deletions(-)

diff --git a/comfy/audio_encoders/audio_encoders.py b/comfy/audio_encoders/audio_encoders.py
index b452ff447..d1ec78f69 100644
--- a/comfy/audio_encoders/audio_encoders.py
+++ b/comfy/audio_encoders/audio_encoders.py
@@ -7,7 +7,7 @@ import torchaudio
 
 
 class AudioEncoderModel():
-    def __init__(self, config, fps=50):
+    def __init__(self, config):
         self.load_device = comfy.model_management.text_encoder_device()
         offload_device = comfy.model_management.text_encoder_offload_device()
         self.dtype = comfy.model_management.text_encoder_dtype(self.load_device)
@@ -21,7 +21,6 @@ class AudioEncoderModel():
         self.model.eval()
         self.patcher = comfy.model_patcher.ModelPatcher(self.model, load_device=self.load_device, offload_device=offload_device)
         self.model_sample_rate = 16000
-        self.fps = fps
 
     def load_sd(self, sd):
         return self.model.load_state_dict(sd, strict=False)
@@ -32,7 +31,7 @@ class AudioEncoderModel():
     def encode_audio(self, audio, sample_rate):
         comfy.model_management.load_model_gpu(self.patcher)
         audio = torchaudio.functional.resample(audio, sample_rate, self.model_sample_rate)
-        out, all_layers = self.model(audio.to(self.load_device), fps=self.fps, sr=self.model_sample_rate)
+        out, all_layers = self.model(audio.to(self.load_device), sr=self.model_sample_rate)
         outputs = {}
         outputs["encoded_audio"] = out
         outputs["encoded_audio_all_layers"] = all_layers
@@ -52,7 +51,6 @@ def load_audio_encoder_from_sd(sd, prefix=""):
             "do_normalize": True,
             "do_stable_layer_norm": True
             }
-        fps = 50
     elif embed_dim == 768: # base
         config = {
             "embed_dim": 768,
@@ -63,11 +61,10 @@ def load_audio_encoder_from_sd(sd, prefix=""):
             "do_normalize": False, # chinese-wav2vec2-base has this False
             "do_stable_layer_norm": False
         }
-        fps = 25
     else:
         raise RuntimeError("ERROR: audio encoder file is invalid or unsupported embed_dim: {}".format(embed_dim))
 
-    audio_encoder = AudioEncoderModel(config, fps=fps)
+    audio_encoder = AudioEncoderModel(config)
     m, u = audio_encoder.load_sd(sd)
     if len(m) > 0:
         logging.warning("missing audio encoder: {}".format(m))
diff --git a/comfy/audio_encoders/wav2vec2.py b/comfy/audio_encoders/wav2vec2.py
index 4933b7501..ef10dcd2a 100644
--- a/comfy/audio_encoders/wav2vec2.py
+++ b/comfy/audio_encoders/wav2vec2.py
@@ -238,26 +238,15 @@ class Wav2Vec2Model(nn.Module):
             device=device, dtype=dtype, operations=operations
         )
 
-    def forward(self, x, fps=50, sr=16000, mask_time_indices=None, return_dict=False):
+    def forward(self, x, sr=16000, mask_time_indices=None, return_dict=False):
         x = torch.mean(x, dim=1)
 
         if self.do_normalize:
             x = (x - x.mean()) / torch.sqrt(x.var() + 1e-7)
 
         features = self.feature_extractor(x)
-
-        if fps != 50:
-            audio_duration = x.shape[1] / sr
-            target_seq_len = int(audio_duration * fps)
-            features = self.linear_interpolation(features, target_seq_len)
-
         features = self.feature_projection(features)
         batch_size, seq_len, _ = features.shape
 
         x, all_x = self.encoder(features)
         return x, all_x
-
-    def linear_interpolation(self, features, target_seq_len):
-        features = features.transpose(1, 2)
-        output_features = torch.nn.functional.interpolate(features, size=target_seq_len, align_corners=True, mode='linear')
-        return output_features.transpose(1, 2)