Instructions to use ncoder-ai/VibeVoice-Large-AWQ-INT4 with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use ncoder-ai/VibeVoice-Large-AWQ-INT4 with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("text-to-speech", model="ncoder-ai/VibeVoice-Large-AWQ-INT4")# Load model directly from transformers import AutoTokenizer, AutoModelForCausalLM tokenizer = AutoTokenizer.from_pretrained("ncoder-ai/VibeVoice-Large-AWQ-INT4") model = AutoModelForCausalLM.from_pretrained("ncoder-ai/VibeVoice-Large-AWQ-INT4") - VibeVoice
How to use ncoder-ai/VibeVoice-Large-AWQ-INT4 with VibeVoice:
import torch, soundfile as sf, librosa, numpy as np from vibevoice.processor.vibevoice_processor import VibeVoiceProcessor from vibevoice.modular.modeling_vibevoice_inference import VibeVoiceForConditionalGenerationInference # Load voice sample (should be 24kHz mono) voice, sr = sf.read("path/to/voice_sample.wav") if voice.ndim > 1: voice = voice.mean(axis=1) if sr != 24000: voice = librosa.resample(voice, sr, 24000) processor = VibeVoiceProcessor.from_pretrained("ncoder-ai/VibeVoice-Large-AWQ-INT4") model = VibeVoiceForConditionalGenerationInference.from_pretrained( "ncoder-ai/VibeVoice-Large-AWQ-INT4", torch_dtype=torch.bfloat16 ).to("cuda").eval() model.set_ddpm_inference_steps(5) inputs = processor(text=["Speaker 0: Hello!\nSpeaker 1: Hi there!"], voice_samples=[[voice]], return_tensors="pt") audio = model.generate(**inputs, cfg_scale=1.3, tokenizer=processor.tokenizer).speech_outputs[0] sf.write("output.wav", audio.cpu().numpy().squeeze(), 24000) - Notebooks
- Google Colab
- Kaggle
Use ncoder-ai/VibeVoice-Large-AWQ instead
This repo holds the AWQ-INT4 Qwen2 LLM weights only, in isolation. It exists so the AWQ-quantized LLM can be composed by hand with a custom VibeVoice base (e.g. a fork, a fine-tune, or a different audio stack).
You almost certainly want the unified drop-in instead: ncoder-ai/VibeVoice-Large-AWQ.
That repo bundles the same AWQ-INT4 LLM with FP16 audio components into one
checkpoint — transformers.from_pretrained() loads it directly, no manual
graft step. Same speed, same VRAM (~8.4 GB), same audio quality.
from vibevoice.modular.modeling_vibevoice_inference import VibeVoiceForConditionalGenerationInference
import torch
model = VibeVoiceForConditionalGenerationInference.from_pretrained(
"ncoder-ai/VibeVoice-Large-AWQ",
torch_dtype=torch.float16,
device_map="cuda:0",
).eval()
If you really need the LLM-only weights
For advanced users hand-grafting the AWQ Qwen2 into a custom base. You provide the FP16 audio stack (acoustic tokenizer, diffusion head, connectors); this repo provides only the quantized language model.
import torch, gc
from vibevoice.modular.modeling_vibevoice_inference import VibeVoiceForConditionalGenerationInference
from awq import AutoAWQForCausalLM
# Your custom FP16 base
model = VibeVoiceForConditionalGenerationInference.from_pretrained(
"rsxdalv/VibeVoice-Large", torch_dtype=torch.float16, device_map="cuda:0",
).eval()
# Free FP16 LLM, graft AWQ Qwen2 in its place
del model.model.language_model
gc.collect(); torch.cuda.empty_cache()
awq = AutoAWQForCausalLM.from_quantized(
"ncoder-ai/VibeVoice-Large-AWQ-INT4",
device_map={"": 0}, safetensors=True, fuse_layers=False,
)
model.model.language_model = awq.model.model
del awq; gc.collect(); torch.cuda.empty_cache()
Quantization recipe: AutoAWQ, 4-bit, group_size=128, GEMM (Marlin) version, zero_point=True. Calibration: 250 samples (200 prose + 50 wikitext), 512-token max length.
License
MIT — same as upstream rsxdalv/VibeVoice-Large.
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Model tree for ncoder-ai/VibeVoice-Large-AWQ-INT4
Base model
rsxdalv/VibeVoice-Large