Model reference · open weights
BUD-E-Whisper is an open-weight audio or speech model from laion. BUD-E-Whisper_V1.2 (FP32) weighs 483 MB; the smallest configuration that runs it is RTX 3060 12 GB.
For self-hosting, compare this build's requirements below with the GPU you plan to rent.
Summary of the laion/BUD-E-Whisper_V1.2 model card, 2026-10-01
What it is
| Released by | laion |
|---|---|
| Released | 2026-04-02 |
| Parameters | 242M |
| VRAM | 483 MB for the weights |
What it runs on
| Card | One stream | Memory |
|---|---|---|
| RTX 3060 12 GB | fits | 11.6 GB |
| RTX 4060 Ti 16 GB | fits | 15.4 GB |
| RTX 3090 24 GB | fits | 23.4 GB |
| RTX 4090 24 GB | fits | 23.4 GB |
| RTX 5090 32 GB | fits | 31.0 GB |
| L40S 48 GB | fits | 44.0 GB |
| A100 80 GB | fits | 78.2 GB |
| H100 80 GB | fits | 78.1 GB |
| RTX PRO 6000 Blackwell 96 GB | fits | 93.8 GB |
| DGX Spark (GB10) 128 GB unified | fits | 107 GB |
| H200 141 GB | fits | 138 GB |
| B200 180 GB | fits | 176 GB |
From the model card
Detailed audio captioning model that generates rich, temporal descriptions of speech audio — including voice characteristics, emotional states, recording quality, speaker demographics, and delivery style.
Fine-tuned from laion/BUD-E-Whisper_V1.1 on the majestrino-unified-detailed-captions-temporal dataset.
laion/BUD-E-Whisper_V1.1 (itself fine-tuned from openai/whisper-small)TTS-AGI/majestrino-unified-detailed-captions-temporal (826 webdataset shards)Given an audio clip (up to 30 seconds), the model generates detailed captions describing:
from transformers import WhisperProcessor, WhisperForConditionalGeneration
import torchaudio
import torch
# Load model
processor = WhisperProcessor.from_pretrained("laion/BUD-E-Whisper_V1.2")
model = WhisperForConditionalGeneration.from_pretrained("laion/BUD-E-Whisper_V1.2")
model.generation_config.forced_decoder_ids = None
model.eval().to("cuda")
# Load audio (resample to 16kHz mono)
wav, sr = torchaudio.load("audio.wav")
if wav.shape[0] > 1:
wav = wav.mean(dim=0, keepdim=True)
if sr != 16000:
wav = torchaudio.functional.resample(wav, sr, 16000)
audio = wav.squeeze(0).numpy()
# Generate caption
inputs = processor.feature_extractor(audio, sampling_rate=16000, return_tensors="pt").to("cuda")
with torch.no_grad():
generated_ids = model.generate(**inputs, max_length=448)
caption = processor.tokenizer.batch_decode(generated_ids, skip_special_tokens=True)[0]
print(caption)
python inference.py audio.wav
python inference.py audio.mp3 --device cpu
Input: A 5-second clip of a male speaker reading a sentence.
Output:
This recording features a clearly masculine adult male speaker delivering a narration with a strong sense of interest and concentration, occasionally exhibiting a hint of contemplation. The overall emotional valence remains neutral, with a slightly calm arousal level throughout. The speaker's demeanor is balanced, displaying a moderate degree of vulnerability alongside a subtle confidence. The vocal delivery is consistently natural and spontaneous, characterized by a neutral pitch and volume, and a smooth, clear timbre with minimal breathiness.
| Parameter | Value |
|---|---|
| Base model | laion/BUD-E-Whisper_V1.1 |
| Dataset | TTS-AGI/majestrino-unified-detailed-captions-temporal |
| Samples trained | ~9M |
| Batch size | 10 per GPU x 2 GPUs = 20 |
| Learning rate | 1e-5 (linear warmup 5%, linear decay) |
| Precision | fp16 mixed |
| Max audio length | 30 seconds |
| Max label tokens | 448 |
| Hardware | 2x NVIDIA RTX 3090 |
| Samples | Val Loss |
|---|---|
| 200 | 3.69 |
| 1M | 0.94 |
| 2M | 0.90 |
| 3M | 0.87 |
| 4M | 0.85 |
| 5M | 0.84 |
| 6M | 0.83 |
| 7M | 0.82 |
| 8M | 0.81 |
| 9M | 0.81 |
@misc{bud-e-whisper-v1.2,
title={BUD-E-Whisper V1.2: Detailed Audio Captioning},
author={LAION},
year={2026},
url={https://huggingface.co/laion/BUD-E-Whisper_V1.2}
}
Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.