去除了torchaudio兼容不行的问题

indextts/infer_v2.py

@@ -6,8 +6,9 @@ import json
 import re
 import time
 import librosa
+import soundfile as sf
+import numpy as np
 import torch
-import torchaudio
 from torch.nn.utils.rnn import pad_sequence
 
 import warnings
@@ -353,6 +354,41 @@ class IndexTTS2:
 
         return emo_vector
 
+    def _compute_fbank_librosa(self, audio, n_mels=80, sr=16000):
+        """
+        使用 librosa 计算 fbank 特征（替代 torchaudio.compliance.kaldi.fbank）
+        兼容 ARM 架构
+        """
+        # 将 torch tensor 转为 numpy
+        device = audio.device
+        audio_np = audio.squeeze().cpu().numpy()
+        
+        # 计算 mel spectrogram（使用与 Kaldi 类似的参数）
+        # n_fft=512, hop_length=160, win_length=400 对应 Kaldi 默认的 25ms window, 10ms shift @ 16kHz
+        mel_spec = librosa.feature.melspectrogram(
+            y=audio_np,
+            sr=sr,
+            n_fft=512,
+            hop_length=160,
+            win_length=400,
+            n_mels=n_mels,
+            fmin=20,
+            fmax=7600,
+            center=False,
+            power=2.0,  # 功率谱
+        )
+        
+        # 转回 torch tensor 并移到原设备
+        mel_spec = torch.from_numpy(mel_spec).to(device)
+        
+        # 应用 log 压缩（Kaldi fbank 默认行为）
+        mel_spec = torch.log(mel_spec + 1e-10)
+        
+        # 转置为 (time, n_mels) 格式，与 Kaldi fbank 输出一致
+        mel_spec = mel_spec.transpose(0, 1)
+        
+        return mel_spec
+
     # 原始推理模式
     def infer(self, spk_audio_prompt, text, output_path,
               emo_audio_prompt=None, emo_alpha=1.0,
@@ -433,8 +469,12 @@ class IndexTTS2:
                 self.cache_mel = None
                 torch.cuda.empty_cache()
             audio,sr = self._load_and_cut_audio(spk_audio_prompt,15,verbose)
-            audio_22k = torchaudio.transforms.Resample(sr, 22050)(audio)
-            audio_16k = torchaudio.transforms.Resample(sr, 16000)(audio)
+            # 使用 librosa 进行重采样
+            audio_np = audio.squeeze().numpy()
+            audio_22k_np = librosa.resample(audio_np, orig_sr=sr, target_sr=22050)
+            audio_16k_np = librosa.resample(audio_np, orig_sr=sr, target_sr=16000)
+            audio_22k = torch.from_numpy(audio_22k_np).unsqueeze(0)
+            audio_16k = torch.from_numpy(audio_16k_np).unsqueeze(0)
 
             inputs = self.extract_features(audio_16k, sampling_rate=16000, return_tensors="pt")
             input_features = inputs["input_features"]
@@ -446,10 +486,8 @@ class IndexTTS2:
             _, S_ref = self.semantic_codec.quantize(spk_cond_emb)
             ref_mel = self.mel_fn(audio_22k.to(spk_cond_emb.device).float())
             ref_target_lengths = torch.LongTensor([ref_mel.size(2)]).to(ref_mel.device)
-            feat = torchaudio.compliance.kaldi.fbank(audio_16k.to(ref_mel.device),
-                                                     num_mel_bins=80,
-                                                     dither=0,
-                                                     sample_frequency=16000)
+            # 使用 librosa 计算 fbank 特征（替代 torchaudio.compliance.kaldi.fbank）
+            feat = self._compute_fbank_librosa(audio_16k.to(ref_mel.device), n_mels=80, sr=16000)
             feat = feat - feat.mean(dim=0, keepdim=True)  # feat2另外一个滤波器能量组特征[922, 80]
             style = self.campplus_model(feat.unsqueeze(0))  # 参考音频的全局style2[1,192]
 
@@ -694,7 +732,9 @@ class IndexTTS2:
                 print(">> remove old wav file:", output_path)
             if os.path.dirname(output_path) != "":
                 os.makedirs(os.path.dirname(output_path), exist_ok=True)
-            torchaudio.save(output_path, wav.type(torch.int16), sampling_rate)
+            # 使用 soundfile 保存音频（替代 torchaudio.save）
+            wav_np = wav.squeeze().numpy().astype(np.int16)
+            sf.write(output_path, wav_np, sampling_rate, subtype='PCM_16')
             print(">> wav file saved to:", output_path)
             if stream_return:
                 return None