FX: Raising the Python version to avoid issues with Gradio failing to start.

Add a streaming API for VoxCPM

.gitignore

@@ -0,0 +1,3 @@
+launch.json
+__pycache__
+voxcpm.egg-info

README.md

@@ -50,7 +50,7 @@ By default, when you first run the script, the model will be downloaded automati
 - Download VoxCPM-0.5B
     ```
     from huggingface_hub import snapshot_download
-    snapshot_download("openbmb/VoxCPM-0.5B",local_files_only=local_files_only)
+    snapshot_download("openbmb/VoxCPM-0.5B")
     ```
 - Download ZipEnhancer and SenseVoice-Small. We use ZipEnhancer to enhance speech prompts and SenseVoice-Small for speech prompt ASR in the web demo.
     ```
@@ -62,10 +62,12 @@ By default, when you first run the script, the model will be downloaded automati
 ### 2. Basic Usage
 ```python
 import soundfile as sf
+import numpy as np
 from voxcpm import VoxCPM
 
 model = VoxCPM.from_pretrained("openbmb/VoxCPM-0.5B")
 
+# Non-streaming
 wav = model.generate(
     text="VoxCPM is an innovative end-to-end TTS model from ModelBest, designed to generate highly expressive speech.",
     prompt_wav_path=None,      # optional: path to a prompt speech for voice cloning
@@ -81,6 +83,18 @@ wav = model.generate(
 
 sf.write("output.wav", wav, 16000)
 print("saved: output.wav")
+
+# Streaming
+chunks = []
+for chunk in model.generate_streaming(
+    text = "Streaming text to speech is easy with VoxCPM!",
+    # supports same args as above
+):
+    chunks.append(chunk)
+wav = np.concatenate(chunks)
+
+sf.write("output_streaming.wav", wav, 16000)
+print("saved: output_streaming.wav")
 ```
 
 ### 3. CLI Usage
@@ -98,6 +112,13 @@ voxcpm --text "VoxCPM is an innovative end-to-end TTS model from ModelBest, desi
   --output out.wav \
   --denoise
 
+# (Optinal) Voice cloning (reference audio + transcript file)
+voxcpm --text "VoxCPM is an innovative end-to-end TTS model from ModelBest, designed to generate highly expressive speech." \
+  --prompt-audio path/to/voice.wav \
+  --prompt-file "/path/to/text-file" \
+  --output out.wav \
+  --denoise
+
 # 3) Batch processing (one text per line)
 voxcpm --input examples/input.txt --output-dir outs
 # (optional) Batch + cloning
@@ -267,6 +288,8 @@ 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

@@ -20,12 +20,10 @@ classifiers = [
     "Intended Audience :: Developers",
     "Operating System :: OS Independent",
     "Programming Language :: Python :: 3",
-    "Programming Language :: Python :: 3.8",
-    "Programming Language :: Python :: 3.9",
     "Programming Language :: Python :: 3.10",
     "Programming Language :: Python :: 3.11",
 ]
-requires-python = ">=3.8"
+requires-python = ">=3.10"
 dependencies = [
     "torch>=2.5.0",
     "torchaudio>=2.5.0",
@@ -36,7 +34,7 @@ dependencies = [
     "addict",
     "wetext",
     "modelscope>=1.22.0",
-    "datasets>=2,<4",
+    "datasets>=3,<4",
     "huggingface-hub",
     "pydantic",
     "tqdm",
@@ -78,7 +76,7 @@ version_scheme = "post-release"
 
 [tool.black]
 line-length = 120
-target-version = ['py38']
+target-version = ['py310']
 include = '\.pyi?$'
 extend-exclude = '''
 /(

src/voxcpm/cli.py

@@ -240,6 +240,7 @@ Examples:
     # Prompt audio (for voice cloning)
     parser.add_argument("--prompt-audio", "-pa", help="Reference audio file path")
     parser.add_argument("--prompt-text", "-pt", help="Reference text corresponding to the audio")
+    parser.add_argument("--prompt-file", "-pf", help="Reference text file corresponding to the audio")
     parser.add_argument("--denoise", action="store_true", help="Enable prompt speech enhancement (denoising)")
 
     # Generation parameters
@@ -279,6 +280,12 @@ def main():
 
     # If prompt audio+text provided → voice cloning
     if args.prompt_audio or args.prompt_text:
+        if not args.prompt_text and args.prompt_file:
+            assert os.path.isfile(args.prompt_file), "Prompt file does not exist or is not accessible."
+        
+            with open(args.prompt_file, 'r', encoding='utf-8') as f:
+                args.prompt_text = f.read()
+
         if not args.prompt_audio or not args.prompt_text:
             print("Error: Voice cloning requires both --prompt-audio and --prompt-text")
             sys.exit(1)

src/voxcpm/core.py

@@ -1,7 +1,8 @@
-import torch
-import torchaudio
 import os
+import re
 import tempfile
+import numpy as np
+from typing import Generator
 from huggingface_hub import snapshot_download
 from .model.voxcpm import VoxCPMModel
 
@@ -10,6 +11,7 @@ class VoxCPM:
             voxcpm_model_path : str,
             zipenhancer_model_path : str = "iic/speech_zipenhancer_ans_multiloss_16k_base",
             enable_denoiser : bool = True,
+            optimize: bool = True,
         ):
         """Initialize VoxCPM TTS pipeline.
 
@@ -20,9 +22,10 @@ class VoxCPM:
             zipenhancer_model_path: ModelScope acoustic noise suppression model
                 id or local path. If None, denoiser will not be initialized.
             enable_denoiser: Whether to initialize the denoiser pipeline.
+            optimize: Whether to optimize the model with torch.compile. True by default, but can be disabled for debugging.
         """
         print(f"voxcpm_model_path: {voxcpm_model_path}, zipenhancer_model_path: {zipenhancer_model_path}, enable_denoiser: {enable_denoiser}")
-        self.tts_model = VoxCPMModel.from_local(voxcpm_model_path)
+        self.tts_model = VoxCPMModel.from_local(voxcpm_model_path, optimize=optimize)
         self.text_normalizer = None
         if enable_denoiser and zipenhancer_model_path is not None:
             from .zipenhancer import ZipEnhancer
@@ -42,6 +45,7 @@ class VoxCPM:
             zipenhancer_model_id: str = "iic/speech_zipenhancer_ans_multiloss_16k_base",
             cache_dir: str = None,
             local_files_only: bool = False,
+            **kwargs,
         ):
         """Instantiate ``VoxCPM`` from a Hugging Face Hub snapshot.
 
@@ -53,6 +57,8 @@ class VoxCPM:
             cache_dir: Custom cache directory for the snapshot.
             local_files_only: If True, only use local files and do not attempt
                 to download.
+        Kwargs:
+            Additional keyword arguments passed to the ``VoxCPM`` constructor.
 
         Returns:
             VoxCPM: Initialized instance whose ``voxcpm_model_path`` points to
@@ -81,9 +87,16 @@ class VoxCPM:
             voxcpm_model_path=local_path,
             zipenhancer_model_path=zipenhancer_model_id if load_denoiser else None,
             enable_denoiser=load_denoiser,
+            **kwargs,
         )
 
-    def generate(self, 
+    def generate(self, *args, **kwargs) -> np.ndarray:
+        return next(self._generate(*args, streaming=False, **kwargs))
+
+    def generate_streaming(self, *args, **kwargs) -> Generator[np.ndarray, None, None]:
+        return self._generate(*args, streaming=True, **kwargs)
+
+    def _generate(self, 
             text : str,
             prompt_wav_path : str = None,
             prompt_text : str = None,
@@ -95,7 +108,8 @@ class VoxCPM:
             retry_badcase : bool = True,
             retry_badcase_max_times : int = 3,
             retry_badcase_ratio_threshold : float = 6.0,
-        ):
+            streaming: bool = False,
+        ) -> Generator[np.ndarray, None, None]:
         """Synthesize speech for the given text and return a single waveform.
 
         This method optionally builds and reuses a prompt cache. If an external
@@ -117,12 +131,24 @@ class VoxCPM:
             retry_badcase: Whether to retry badcase.
             retry_badcase_max_times: Maximum number of times to retry badcase.
             retry_badcase_ratio_threshold: Threshold for audio-to-text ratio.
+            streaming: Whether to return a generator of audio chunks.
         Returns:
-            numpy.ndarray: 1D waveform array (float32) on CPU.
+            Generator of numpy.ndarray: 1D waveform array (float32) on CPU. 
+            Yields audio chunks for each generations step if ``streaming=True``,
+            otherwise yields a single array containing the final audio.
         """
-        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 +165,27 @@ 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()
+            generate_result = 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,
+                            streaming=streaming,
+                        )
+        
+            for wav, _, _ in generate_result:
+                yield wav.squeeze(0).cpu().numpy()
         
         finally:
             if temp_prompt_wav_path and os.path.exists(temp_prompt_wav_path):

src/voxcpm/model/voxcpm.py

@@ -19,11 +19,12 @@ limitations under the License.
 """
 
 import os
-from typing import Dict, Optional, Tuple, Union
+from typing import Tuple, Union, Generator, List
 
 import torch
 import torch.nn as nn
 import torchaudio
+import warnings
 from einops import rearrange
 from pydantic import BaseModel
 from tqdm import tqdm
@@ -147,16 +148,23 @@ class VoxCPMModel(nn.Module):
         self.sample_rate = audio_vae.sample_rate
 
     
-    def optimize(self):
+    def optimize(self, disable: bool = False):
         try:
+            if disable:
+                raise ValueError("Optimization disabled by user")
             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
@@ -164,8 +172,14 @@ class VoxCPMModel(nn.Module):
         return self
 
 
+    def generate(self, *args, **kwargs) -> torch.Tensor:
+        return next(self._generate(*args, streaming=False, **kwargs))
+
+    def generate_streaming(self, *args, **kwargs) -> Generator[torch.Tensor, None, None]:
+        return self._generate(*args, streaming=True, **kwargs)
+
     @torch.inference_mode()
-    def generate(
+    def _generate(
         self,
         target_text: str,
         prompt_text: str = "",
@@ -177,7 +191,11 @@ class VoxCPMModel(nn.Module):
         retry_badcase: bool = False,
         retry_badcase_max_times: int = 3,
         retry_badcase_ratio_threshold: float = 6.0, # setting acceptable ratio of audio length to text length (for badcase detection)
-    ):
+        streaming: bool = False,
+    ) -> Generator[torch.Tensor, None, None]:
+        if retry_badcase and streaming:
+            warnings.warn("Retry on bad cases is not supported in streaming mode, setting retry_badcase=False.")
+            retry_badcase = False
         if len(prompt_wav_path) == 0:
             text = target_text
             text_token = torch.LongTensor(self.text_tokenizer(text))
@@ -260,7 +278,7 @@ class VoxCPMModel(nn.Module):
         
         retry_badcase_times = 0
         while retry_badcase_times < retry_badcase_max_times:
-            latent_pred, pred_audio_feat = self.inference(
+            inference_result = self._inference(
                 text_token,
                 text_mask,
                 audio_feat,
@@ -269,17 +287,31 @@ class VoxCPMModel(nn.Module):
                 max_len=int(target_text_length * retry_badcase_ratio_threshold + 10) if retry_badcase else max_len,
                 inference_timesteps=inference_timesteps,
                 cfg_value=cfg_value,
+                streaming=streaming,
             )
-            if retry_badcase:
-                if pred_audio_feat.shape[0] >= target_text_length * retry_badcase_ratio_threshold:
-                    print(f"  Badcase detected, audio_text_ratio={pred_audio_feat.shape[0] / target_text_length}, retrying...")
-                    retry_badcase_times += 1
-                    continue
+            if streaming:
+                patch_len = self.patch_size * self.chunk_size
+                for latent_pred, _ in inference_result:
+                    decode_audio = self.audio_vae.decode(latent_pred.to(torch.float32))
+                    decode_audio = decode_audio[..., -patch_len:].squeeze(1).cpu()
+                    yield decode_audio
+                break
+            else:
+                latent_pred, pred_audio_feat = next(inference_result)
+                if retry_badcase:
+                    if pred_audio_feat.shape[0] >= target_text_length * retry_badcase_ratio_threshold:
+                        print(f"  Badcase detected, audio_text_ratio={pred_audio_feat.shape[0] / target_text_length}, retrying...")
+                        retry_badcase_times += 1
+                        continue
+                    else:
+                        break
                 else:
                     break   
-            else:
-                break
-        return self.audio_vae.decode(latent_pred.to(torch.float32)).squeeze(1).cpu()
+                
+        if not streaming:
+            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
+            yield decode_audio        
     
     @torch.inference_mode()
     def build_prompt_cache(
@@ -317,7 +349,7 @@ 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,
@@ -369,8 +401,16 @@ class VoxCPMModel(nn.Module):
         
         return merged_cache
 
+    def generate_with_prompt_cache(self, *args, **kwargs) -> Tuple[torch.Tensor, torch.Tensor, torch.Tensor]:
+        return next(self._generate_with_prompt_cache(*args, streaming=False, **kwargs))
+
+    def generate_with_prompt_cache_streaming(
+        self, *args, **kwargs
+    ) -> Generator[Tuple[torch.Tensor, torch.Tensor, List[torch.Tensor]], None, None]:
+        return self._generate_with_prompt_cache(*args, streaming=True, **kwargs)
+
     @torch.inference_mode()
-    def generate_with_prompt_cache(
+    def _generate_with_prompt_cache(
         self,
         target_text: str,
         prompt_cache: dict,
@@ -381,7 +421,8 @@ class VoxCPMModel(nn.Module):
         retry_badcase: bool = False,
         retry_badcase_max_times: int = 3,
         retry_badcase_ratio_threshold: float = 6.0,
-    ):
+        streaming: bool = False,
+    ) -> Generator[Tuple[torch.Tensor, torch.Tensor, Union[torch.Tensor, List[torch.Tensor]]], None, None]:
         """
         Generate audio using pre-built prompt cache.
         
@@ -395,10 +436,17 @@ class VoxCPMModel(nn.Module):
             retry_badcase: Whether to retry on bad cases
             retry_badcase_max_times: Maximum retry attempts
             retry_badcase_ratio_threshold: Threshold for audio-to-text ratio
+            streaming: Whether to return a generator of audio chunks
             
         Returns:
-            tuple: (decoded audio tensor, new text tokens, new audio features)
+            Generator of Tuple containing:
+                - Decoded audio tensor for the current step if ``streaming=True``, else final decoded audio tensor
+                - Tensor of new text tokens
+                - New audio features up to the current step as a List if ``streaming=True``, else as a concatenated Tensor
         """
+        if retry_badcase and streaming:
+            warnings.warn("Retry on bad cases is not supported in streaming mode, setting retry_badcase=False.")
+            retry_badcase = False
         # get prompt from cache
         if prompt_cache is None:
             prompt_text_token = torch.empty(0, dtype=torch.int32)
@@ -443,7 +491,7 @@ class VoxCPMModel(nn.Module):
         target_text_length = len(self.text_tokenizer(target_text))
         retry_badcase_times = 0
         while retry_badcase_times < retry_badcase_max_times:
-            latent_pred, pred_audio_feat = self.inference(
+            inference_result = self._inference(
                 text_token,
                 text_mask,
                 audio_feat,
@@ -452,26 +500,48 @@ class VoxCPMModel(nn.Module):
                 max_len=int(target_text_length * retry_badcase_ratio_threshold + 10) if retry_badcase else max_len,
                 inference_timesteps=inference_timesteps,
                 cfg_value=cfg_value,
+                streaming=streaming,
             )
-            if retry_badcase:
-                if pred_audio_feat.shape[0] >= target_text_length * retry_badcase_ratio_threshold:
-                    print(f"  Badcase detected, audio_text_ratio={pred_audio_feat.shape[0] / target_text_length}, retrying...")
-                    retry_badcase_times += 1
-                    continue
+            if streaming:
+                patch_len = self.patch_size * self.chunk_size
+                for latent_pred, pred_audio_feat in inference_result:
+                    decode_audio = self.audio_vae.decode(latent_pred.to(torch.float32))
+                    decode_audio = decode_audio[..., -patch_len:].squeeze(1).cpu()
+                    yield (
+                        decode_audio,
+                        target_text_token,
+                        pred_audio_feat
+                    )
+                break
+            else:
+                latent_pred, pred_audio_feat = next(inference_result)
+                if retry_badcase:
+                    if pred_audio_feat.shape[0] >= target_text_length * retry_badcase_ratio_threshold:
+                        print(f"  Badcase detected, audio_text_ratio={pred_audio_feat.shape[0] / target_text_length}, retrying...")
+                        retry_badcase_times += 1
+                        continue
+                    else:
+                        break
                 else:
                     break
-            else:
-                break
-        decode_audio = self.audio_vae.decode(latent_pred.to(torch.float32)).squeeze(1).cpu()
+        if not streaming:
+            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,
-            target_text_token,
-            pred_audio_feat
-        )
+            yield (
+                decode_audio,
+                target_text_token,
+                pred_audio_feat
+            )
+
+    def inference(self, *args, **kwargs) -> Tuple[torch.Tensor, torch.Tensor]:
+        return next(self._inference(*args, streaming=False, **kwargs))
+    
+    def inference_streaming(self, *args, **kwargs) -> Generator[Tuple[torch.Tensor, List[torch.Tensor]], None, None]:
+        return self._inference(*args, streaming=True, **kwargs)
 
     @torch.inference_mode()
-    def inference(
+    def _inference(
         self,
         text: torch.Tensor,
         text_mask: torch.Tensor,
@@ -481,7 +551,8 @@ class VoxCPMModel(nn.Module):
         max_len: int = 2000,
         inference_timesteps: int = 10,
         cfg_value: float = 2.0,
-    ) -> Tuple[torch.Tensor, torch.Tensor]:
+        streaming: bool = False,
+    ) -> Generator[Tuple[torch.Tensor, Union[torch.Tensor, List[torch.Tensor]]], None, None]:
         """Core inference method for audio generation.
         
         This is the main inference loop that generates audio features
@@ -496,11 +567,12 @@ class VoxCPMModel(nn.Module):
             max_len: Maximum generation length
             inference_timesteps: Number of diffusion steps
             cfg_value: Classifier-free guidance value
+            streaming: Whether to yield each step latent feature or just the final result
             
         Returns:
-            Tuple containing:
-                - Predicted latent features
-                - Predicted audio feature sequence
+            Generator of Tuple containing:
+                - Predicted latent feature at the current step if ``streaming=True``, else final latent features
+                - Predicted audio feature sequence so far as a List if ``streaming=True``, else as a concatenated Tensor
         """
         B, T, P, D = feat.shape
 
@@ -558,6 +630,12 @@ class VoxCPMModel(nn.Module):
             pred_feat_seq.append(pred_feat.unsqueeze(1))  # b, 1, p, d
             prefix_feat_cond = pred_feat
 
+            if streaming:
+                # return the last three predicted latent features to provide enough context for smooth decoding
+                pred_feat_chunk = torch.cat(pred_feat_seq[-3:], dim=1)
+                feat_pred = rearrange(pred_feat_chunk, "b t p d -> b d (t p)", b=B, p=self.patch_size)
+                yield feat_pred, pred_feat_seq
+            
             stop_flag = self.stop_head(self.stop_actn(self.stop_proj(lm_hidden))).argmax(dim=-1)[0].cpu().item()
             if i > min_len and stop_flag == 1:
                 break
@@ -572,14 +650,14 @@ class VoxCPMModel(nn.Module):
                 lm_hidden + curr_embed[:, 0, :], torch.tensor([self.residual_lm.kv_cache.step()], device=curr_embed.device)
             ).clone()
                 
-        pred_feat_seq = torch.cat(pred_feat_seq, dim=1)  # b, t, p, d
+        if not streaming:
+            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()
+            feat_pred = rearrange(pred_feat_seq, "b t p d -> b d (t p)", b=B, p=self.patch_size)
+            yield feat_pred, pred_feat_seq.squeeze(0).cpu()
 
     @classmethod
-    def from_local(cls, path: str):
+    def from_local(cls, path: str, optimize: bool = True):
         config = VoxCPMConfig.model_validate_json(open(os.path.join(path, "config.json")).read())
 
         tokenizer = LlamaTokenizerFast.from_pretrained(path)
@@ -605,4 +683,4 @@ class VoxCPMModel(nn.Module):
         for kw, val in vae_state_dict.items():
             model_state_dict[f"audio_vae.{kw}"] = val
         model.load_state_dict(model_state_dict, strict=True)
-        return model.to(model.device).eval().optimize()
+        return model.to(model.device).eval().optimize(disable=not optimize)