Model reference · open weights

BUD-E-Whisper

Audio laion Speech→text 1 build Open weights 797 dl/mo

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.

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  • BUD-E-Whisper V1.2 is a 242M parameter automatic speech recognition model developed by laion that generates detailed temporal captions for speech audio.
  • It describes voice characteristics, emotional states, and recording quality for inputs up to 30 seconds long.
  • The model is trained on English-centric data with some multilingual capability and is released under the cc-by-4.0 licence.

Summary of the laion/BUD-E-Whisper_V1.2 model card, 2026-10-01

What it is

Released bylaion
Released2026-04-02
Parameters242M
VRAM483 MB for the weights

What it runs on

Memory and cards for BUD-E-Whisper_V1.2 (FP32)

483 MBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

From the model card

What laion says about BUD-E-Whisper

Read 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.

Model Details

  • Architecture: Whisper Small (encoder-decoder, 242M parameters)
  • Base model: laion/BUD-E-Whisper_V1.1 (itself fine-tuned from openai/whisper-small)
  • Training data: ~9M samples from TTS-AGI/majestrino-unified-detailed-captions-temporal (826 webdataset shards)
  • Training: 2x RTX 3090, DDP (gloo), fp16 mixed precision, AdamW (lr=1e-5, linear warmup 5%), batch size 20
  • Final validation loss: 0.81

What it outputs

Given an audio clip (up to 30 seconds), the model generates detailed captions describing:

  • Speaker demographics: age range, gender, accent
  • Voice timbre: pitch, brightness, breathiness, nasality, resonance
  • Emotional state: valence, arousal, dominance, specific emotions
  • Delivery style: tempo, fluency, expressiveness, naturalness
  • Recording quality: background noise, clarity, studio vs. field
  • Temporal aspects: how delivery and emotion change over time

Quick Start

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)

CLI Inference

python inference.py audio.wav
python inference.py audio.mp3 --device cpu

Example Output

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.

Training Details

ParameterValue
Base modellaion/BUD-E-Whisper_V1.1
DatasetTTS-AGI/majestrino-unified-detailed-captions-temporal
Samples trained~9M
Batch size10 per GPU x 2 GPUs = 20
Learning rate1e-5 (linear warmup 5%, linear decay)
Precisionfp16 mixed
Max audio length30 seconds
Max label tokens448
Hardware2x NVIDIA RTX 3090

Validation Loss Progression

SamplesVal Loss
2003.69
1M0.94
2M0.90
3M0.87
4M0.85
5M0.84
6M0.83
7M0.82
8M0.81
9M0.81

Limitations

  • Optimized for speech audio; may produce less meaningful captions for music or environmental sounds
  • Maximum input length is 30 seconds
  • English-centric training data, though it can handle some other languages (e.g., German)
  • May occasionally hallucinate speaker gender or specific emotional states

Citation

@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.

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