Fx: capture compile error on Windows

README.md

@@ -267,6 +267,8 @@ This project is developed by the following institutions:
 - <img src="assets/thuhcsi_logo.png" width="28px"> [THUHCSI](https://github.com/thuhcsi)
 
 
+## ⭐ Star History
+ [![Star History Chart](https://api.star-history.com/svg?repos=OpenBMB/VoxCPM&type=Date)](https://star-history.com/#OpenBMB/VoxCPM&Date)
 
 
 ## 📚 Citation

app.py

@@ -194,10 +194,6 @@ def create_demo_interface(demo: VoxCPMDemo):
               **调低**：合成速度更快。
             - **Higher** for better synthesis quality.  
               **调高**：合成质量更佳。
-
-            ### Long Text (e.g., >5 min speech)｜长文本 (如 >5分钟的合成语音)
-            While VoxCPM can handle long texts directly, we recommend using empty lines to break very long content into paragraphs; the model will then synthesize each paragraph individually.  
-            虽然 VoxCPM 支持直接生成长文本，但如果目标文本过长，我们建议使用换行符将内容分段；模型将对每个段落分别合成。
             """)
 
         # Main controls
@@ -206,7 +202,7 @@ def create_demo_interface(demo: VoxCPMDemo):
                 prompt_wav = gr.Audio(
                     sources=["upload", 'microphone'],
                     type="filepath",
-                    label="Prompt Speech",
+                    label="Prompt Speech (Optional, or let VoxCPM improvise)",
                     value="./examples/example.wav",
                 )
                 DoDenoisePromptAudio = gr.Checkbox(
@@ -244,14 +240,13 @@ def create_demo_interface(demo: VoxCPMDemo):
                     text = gr.Textbox(
                         value="VoxCPM is an innovative end-to-end TTS model from ModelBest, designed to generate highly realistic speech.",
                         label="Target Text",
-                        info="Default processing splits text on \\n into paragraphs; each is synthesized as a chunk and then concatenated into the final audio."
                     )
                 with gr.Row():
                     DoNormalizeText = gr.Checkbox(
                         value=False,
                         label="Text Normalization",
                         elem_id="chk_normalize",
-                        info="We use WeTextPorcessing library to normalize the input text."
+                        info="We use wetext library to normalize the input text."
                     )
                 audio_output = gr.Audio(label="Output Audio")
 

pyproject.toml

@@ -36,7 +36,7 @@ dependencies = [
     "addict",
     "wetext",
     "modelscope>=1.22.0",
-    "datasets>=2,<4",
+    "datasets>=3,<4",
     "huggingface-hub",
     "pydantic",
     "tqdm",

src/voxcpm/core.py

@@ -1,6 +1,7 @@
 import torch
 import torchaudio
 import os
+import re
 import tempfile
 from huggingface_hub import snapshot_download
 from .model.voxcpm import VoxCPMModel
@@ -120,9 +121,18 @@ class VoxCPM:
         Returns:
             numpy.ndarray: 1D waveform array (float32) on CPU.
         """
-        texts = text.split("\n")
-        texts = [t.strip() for t in texts if t.strip()]
-        final_wav = []
+        if not text.strip() or not isinstance(text, str):
+            raise ValueError("target text must be a non-empty string")
+        
+        if prompt_wav_path is not None:
+            if not os.path.exists(prompt_wav_path):
+                raise FileNotFoundError(f"prompt_wav_path does not exist: {prompt_wav_path}")
+        
+        if (prompt_wav_path is None) != (prompt_text is None):
+            raise ValueError("prompt_wav_path and prompt_text must both be provided or both be None")
+        
+        text = text.replace("\n", " ")
+        text = re.sub(r'\s+', ' ', text)
         temp_prompt_wav_path = None
         
         try:
@@ -139,35 +149,25 @@ class VoxCPM:
             else:
                 fixed_prompt_cache = None  # will be built from the first inference
             
-            for sub_text in texts:
-                if sub_text.strip() == "":
-                    continue
-                print("sub_text:", sub_text)
-                if normalize:
-                    if self.text_normalizer is None:
-                        from .utils.text_normalize import TextNormalizer
-                        self.text_normalizer = TextNormalizer()
-                    sub_text = self.text_normalizer.normalize(sub_text)
-                wav, target_text_token, generated_audio_feat = self.tts_model.generate_with_prompt_cache(
-                                target_text=sub_text,
-                                prompt_cache=fixed_prompt_cache,
-                                min_len=2,
-                                max_len=max_length,
-                                inference_timesteps=inference_timesteps,
-                                cfg_value=cfg_value,
-                                retry_badcase=retry_badcase,
-                                retry_badcase_max_times=retry_badcase_max_times,
-                                retry_badcase_ratio_threshold=retry_badcase_ratio_threshold,
-                            )
-                if fixed_prompt_cache is None:
-                    fixed_prompt_cache = self.tts_model.merge_prompt_cache(
-                        original_cache=None,
-                        new_text_token=target_text_token,
-                        new_audio_feat=generated_audio_feat
-                    )
-                final_wav.append(wav)
+            if normalize:
+                if self.text_normalizer is None:
+                    from .utils.text_normalize import TextNormalizer
+                    self.text_normalizer = TextNormalizer()
+                text = self.text_normalizer.normalize(text)
             
-            return torch.cat(final_wav, dim=1).squeeze(0).cpu().numpy()
+            wav, target_text_token, generated_audio_feat = self.tts_model.generate_with_prompt_cache(
+                            target_text=text,
+                            prompt_cache=fixed_prompt_cache,
+                            min_len=2,
+                            max_len=max_length,
+                            inference_timesteps=inference_timesteps,
+                            cfg_value=cfg_value,
+                            retry_badcase=retry_badcase,
+                            retry_badcase_max_times=retry_badcase_max_times,
+                            retry_badcase_ratio_threshold=retry_badcase_ratio_threshold,
+                        )
+        
+            return wav.squeeze(0).cpu().numpy()
         
         finally:
             if temp_prompt_wav_path and os.path.exists(temp_prompt_wav_path):

src/voxcpm/model/voxcpm.py

@@ -151,12 +151,17 @@ class VoxCPMModel(nn.Module):
         try:
             if self.device != "cuda":
                 raise ValueError("VoxCPMModel can only be optimized on CUDA device")
+            try:
+                import triton
+            except:
+                raise ValueError("triton is not installed")
             self.base_lm.forward_step = torch.compile(self.base_lm.forward_step, mode="reduce-overhead", fullgraph=True)
             self.residual_lm.forward_step = torch.compile(self.residual_lm.forward_step, mode="reduce-overhead", fullgraph=True)
             self.feat_encoder_step = torch.compile(self.feat_encoder, mode="reduce-overhead", fullgraph=True)
             self.feat_decoder.estimator = torch.compile(self.feat_decoder.estimator, mode="reduce-overhead", fullgraph=True)
-        except:
-            print("VoxCPMModel can not be optimized by torch.compile, using original forward_step functions")
+        except Exception as e:
+            print(f"Error: {e}")
+            print("Warning: VoxCPMModel can not be optimized by torch.compile, using original forward_step functions")
             self.base_lm.forward_step = self.base_lm.forward_step
             self.residual_lm.forward_step = self.residual_lm.forward_step
             self.feat_encoder_step = self.feat_encoder
@@ -279,7 +284,10 @@ class VoxCPMModel(nn.Module):
                     break
             else:
                 break   
-        return self.audio_vae.decode(latent_pred.to(torch.float32)).squeeze(1).cpu()
+                
+        decode_audio = self.audio_vae.decode(latent_pred.to(torch.float32)).squeeze(1).cpu()  
+        decode_audio = decode_audio[..., 640:-640] # trick: trim the start and end of the audio
+        return decode_audio        
     
     @torch.inference_mode()
     def build_prompt_cache(
@@ -317,7 +325,7 @@ class VoxCPMModel(nn.Module):
             audio = torch.nn.functional.pad(audio, (0, patch_len - audio.size(1) % patch_len))
 
         # extract audio features
-        audio_feat = self.audio_vae.encode(audio.cuda(), self.sample_rate).cpu()
+        audio_feat = self.audio_vae.encode(audio.to(self.device), self.sample_rate).cpu()
 
         audio_feat = audio_feat.view(
             self.audio_vae.latent_dim,
@@ -463,6 +471,7 @@ class VoxCPMModel(nn.Module):
             else:
                 break
         decode_audio = self.audio_vae.decode(latent_pred.to(torch.float32)).squeeze(1).cpu()
+        decode_audio = decode_audio[..., 640:-640] # trick: trim the start and end of the audio
 
         return (
             decode_audio,
@@ -575,7 +584,6 @@ class VoxCPMModel(nn.Module):
         pred_feat_seq = torch.cat(pred_feat_seq, dim=1)  # b, t, p, d
 
         feat_pred = rearrange(pred_feat_seq, "b t p d -> b d (t p)", b=B, p=self.patch_size)
-        feat_pred = feat_pred[..., 1:-1] # trick: remove the first and last token
         return feat_pred, pred_feat_seq.squeeze(0).cpu()
 
     @classmethod