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bosonai/higgs-audio-v3-8b-stt

sourceHugging Faceapache-2.0updated 4mo agoView on Hugging Face
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Higgs Audio v3 8B STT

A speech-to-text model combining a Whisper-Large-v3 encoder with a Qwen3-8B decoder (8.91B total parameters), fine-tuned with LoRA on diverse ASR benchmarks.

Usage

python
import torch
import numpy as np
from transformers import AutoModel, AutoTokenizer

# Load model
model = AutoModel.from_pretrained(
    "bosonai/higgs-audio-v3-8b-stt",
    torch_dtype=torch.bfloat16,
    trust_remote_code=True,
    attn_implementation="eager",
    device_map="cuda:0",
)
tokenizer = AutoTokenizer.from_pretrained("bosonai/higgs-audio-v3-8b-stt")

# Transcribe audio (16kHz mono numpy array)
from transformers.utils import cached_file
import importlib.util
spec = importlib.util.spec_from_file_location("transcribe", cached_file("bosonai/higgs-audio-v3-8b-stt", "transcribe.py", _raise_exceptions_for_connection_errors=False))
mod = importlib.util.module_from_spec(spec)
spec.loader.exec_module(mod)

audio_np = np.random.randn(16000).astype(np.float32)  # replace with your audio
text = mod.transcribe(model, tokenizer, audio_np)
print(text)

Requirements

torch
transformers>=4.51.0
whisper  # for audio preprocessing (WhisperProcessor)

Architecture

  • —Encoder: Whisper-Large-v3 (frozen)
  • —Decoder: Qwen3-8B (LoRA fine-tuned, merged)
  • —Total parameters: 8.91B
  • —Audio input: 16kHz mono WAV
  • —Supports: Thinking mode for improved accuracy

Performance (ESB Benchmark — Full Scale, All Samples)

DatasetWER
AMI6.23%
Earnings2211.33%
GigaSpeech9.34%
LibriSpeech Clean1.24%
LibriSpeech Other2.34%
SPGISpeech3.14%
TED-LIUM3.14%
VoxPopuli5.63%
Average5.30%

Training Data

  • —10K AMI samples
  • —6K SPGISpeech
  • —5K Earnings22
  • —4K VoxPopuli
  • —3K LibriSpeech
  • —3K TED-LIUM