bosonai/higgs-audio-v3-8b-stt-v2

Model

Higgs Audio v3 8B STT v2

15

6 commits

1 linked in READMEs

updated May 22, 2026

See the code

README

Higgs Audio v3 8B STT v2

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

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

# Load model
model = AutoModel.from_pretrained(
    "bosonai/higgs-audio-v3-8b-stt-v2",
    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-v2")

# 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-v2", "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
AMI10.14%
Earnings228.73%
GigaSpeech8.47%
LibriSpeech Clean1.25%
LibriSpeech Other2.38%
SPGISpeech3.60%
TED-LIUM3.09%
VoxPopuli5.92%
Average5.449%
automatic-speech-recognition
custom_code
hf-asr-leaderboard
higgs_audio_3
model-index
qwen
safetensors
whisper

Contributors

erik-at-boson

5 commits

aimesh-repo

1 commits

bosonai/higgs-audio-v3-8b-stt-v2

Model

Higgs Audio v3 8B STT v2

15

6 commits

1 linked in READMEs

updated May 22, 2026

See the code

README

Higgs Audio v3 8B STT v2

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

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

# Load model
model = AutoModel.from_pretrained(
    "bosonai/higgs-audio-v3-8b-stt-v2",
    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-v2")

# 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-v2", "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
AMI10.14%
Earnings228.73%
GigaSpeech8.47%
LibriSpeech Clean1.25%
LibriSpeech Other2.38%
SPGISpeech3.60%
TED-LIUM3.09%
VoxPopuli5.92%
Average5.449%
automatic-speech-recognition
custom_code
hf-asr-leaderboard
higgs_audio_3
model-index
qwen
safetensors
whisper

Contributors

erik-at-boson

5 commits

aimesh-repo

1 commits