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Running
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Running
on
CPU Upgrade
Commit
·
93a624e
1
Parent(s):
34aa391
Revert from transformer version
Browse files- app.py +29 -13
- requirements.txt +9 -2
app.py
CHANGED
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@@ -6,10 +6,13 @@ import librosa
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import soundfile as sf
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import torch
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import traceback
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from spaces import GPU
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from datetime import datetime
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from
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from transformers.utils import logging
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from transformers import set_seed
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@@ -44,6 +47,17 @@ class VibeVoiceDemo:
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def load_models(self):
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print("Loading processors and models on CPU...")
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for name, path in self.model_paths.items():
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print(f" - {name} from {path}")
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proc = VibeVoiceProcessor.from_pretrained(path)
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@@ -159,7 +173,15 @@ class VibeVoiceDemo:
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log += f"Parameters: CFG Scale={cfg_scale}\n"
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log += f"Speakers: {', '.join(selected_speakers)}\n"
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-
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lines = script.strip().split('\n')
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formatted_script_lines = []
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@@ -177,18 +199,13 @@ class VibeVoiceDemo:
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log += f"Formatted script with {len(formatted_script_lines)} turns\n"
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log += "Processing with VibeVoice...\n"
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# Processor now expects file paths, not audio arrays
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voice_sample_paths = [self.available_voices[speaker] for speaker in selected_speakers]
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-
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inputs = processor(
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text=[formatted_script],
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voice_samples=[
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padding=True,
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return_tensors="pt",
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)
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# Move inputs to device
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inputs = {key: val.to(self.device) if isinstance(val, torch.Tensor) else val for key, val in inputs.items()}
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start_time = time.time()
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outputs = model.generate(
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@@ -196,7 +213,7 @@ class VibeVoiceDemo:
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max_new_tokens=None,
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cfg_scale=cfg_scale,
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tokenizer=processor.tokenizer,
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generation_config={'do_sample': False
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verbose=False,
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)
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generation_time = time.time() - start_time
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@@ -210,8 +227,7 @@ class VibeVoiceDemo:
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if audio.ndim > 1:
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audio = audio.squeeze()
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sample_rate = processor.audio_processor.sampling_rate if hasattr(processor, 'audio_processor') else 24000
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output_dir = "outputs"
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os.makedirs(output_dir, exist_ok=True)
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@@ -714,7 +730,7 @@ def run_demo(
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model_paths: dict = None,
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device: str = "cuda",
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inference_steps: int = 5,
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share: bool =
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):
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"""
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model_paths default includes two entries. Replace paths as needed.
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import soundfile as sf
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import torch
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import traceback
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import threading
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from spaces import GPU
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from datetime import datetime
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from modular.modeling_vibevoice_inference import VibeVoiceForConditionalGenerationInference
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from processor.vibevoice_processor import VibeVoiceProcessor
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from modular.streamer import AudioStreamer
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from transformers.utils import logging
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from transformers import set_seed
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def load_models(self):
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print("Loading processors and models on CPU...")
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# Debug: Show cache location
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import os
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cache_dir = os.path.expanduser("~/.cache/huggingface/hub")
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print(f"HuggingFace cache directory: {cache_dir}")
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if os.path.exists(cache_dir):
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print(f"Cache exists. Size: {sum(os.path.getsize(os.path.join(dirpath, filename)) for dirpath, _, filenames in os.walk(cache_dir) for filename in filenames) / (1024**3):.2f} GB")
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print("Cached models:")
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for item in os.listdir(cache_dir):
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if item.startswith("models--"):
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print(f" - {item}")
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for name, path in self.model_paths.items():
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print(f" - {name} from {path}")
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proc = VibeVoiceProcessor.from_pretrained(path)
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log += f"Parameters: CFG Scale={cfg_scale}\n"
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log += f"Speakers: {', '.join(selected_speakers)}\n"
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voice_samples = []
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for speaker_name in selected_speakers:
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audio_path = self.available_voices[speaker_name]
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audio_data = self.read_audio(audio_path)
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if len(audio_data) == 0:
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raise gr.Error(f"Error: Failed to load audio for {speaker_name}")
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voice_samples.append(audio_data)
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log += f"Loaded {len(voice_samples)} voice samples\n"
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lines = script.strip().split('\n')
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formatted_script_lines = []
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log += f"Formatted script with {len(formatted_script_lines)} turns\n"
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log += "Processing with VibeVoice...\n"
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inputs = processor(
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text=[formatted_script],
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voice_samples=[voice_samples],
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padding=True,
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return_tensors="pt",
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return_attention_mask=True,
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)
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start_time = time.time()
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outputs = model.generate(
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max_new_tokens=None,
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cfg_scale=cfg_scale,
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tokenizer=processor.tokenizer,
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generation_config={'do_sample': False},
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verbose=False,
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)
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generation_time = time.time() - start_time
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if audio.ndim > 1:
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audio = audio.squeeze()
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sample_rate = 24000
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output_dir = "outputs"
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os.makedirs(output_dir, exist_ok=True)
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model_paths: dict = None,
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device: str = "cuda",
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inference_steps: int = 5,
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share: bool = False,
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):
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"""
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model_paths default includes two entries. Replace paths as needed.
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requirements.txt
CHANGED
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@@ -1,12 +1,19 @@
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spaces
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torch
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accelerate
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-
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diffusers
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tqdm
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numpy
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scipy
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gradio
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soundfile
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librosa
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spaces
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torch
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accelerate==1.6.0
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transformers==4.51.3
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diffusers
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tqdm
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numpy
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scipy
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ml-collections
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absl-py
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gradio
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av
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aiortc
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soundfile
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librosa
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pydub
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requests
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python-dotenv
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