Model reference · open weights
granite-4.0-speech is an open-weight audio or speech model from IBM. granite-4.0-1b-speech (BF16) weighs 4.6 GB; the smallest configuration that runs it is RTX 3060 12 GB.
Summary of the ibm-granite/granite-4.0-1b-speech model card, 2026-10-01
What it is
| Released by | IBM |
|---|---|
| Released | 2026-02-27 |
| Parameters | 2.3B |
| VRAM | 4.6 GB 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
Model Summary: Granite-4.0-1b-speech is a compact and efficient speech-language model, specifically designed for multilingual automatic speech recognition (ASR) and bidirectional automatic speech translation (AST).
The model was trained on a collection of public corpora comprising of diverse datasets for ASR and AST as well as synthetic datasets tailored to support Japanese ASR, keyword-biased ASR and speech translation. Granite-4.0-1b-speech was trained by modality aligning granite-4.0-1b-base to speech on publicly available open source corpora containing audio inputs and text targets. Compared to granite-speech-3.3-2b and granite-speech-3.3-8b, this model has the following additional capabilities and improvements:
Evaluations:
We evaluated granite-4.0-1b-speech alongside other speech-language models in the less than 8b parameter range as well as dedicated ASR and AST systems on standard benchmarks. The evaluation spanned multiple public benchmarks, with particular emphasis on English ASR tasks while also including multilingual ASR and AST for X-En and En-X translations.
Performance on HuggingFace Open ASR leaderboard:
| model | Average WER | RTFx | AMI | Earnings22 | Gigaspeech | LS Clean | LS Other | SPGISpeech | Tedlium | Voxpopuli |
|---|---|---|---|---|---|---|---|---|---|---|
| ibm-granite/granite-4.0-1b-speech | 5.52 | 280.02 | 8.44 | 8.48 | 10.14 | 1.42 | 2.85 | 3.89 | 3.1 | 5.84 |
Release Date: March 6, 2026
License: Apache 2.0
Supported Languages: English, French, German, Spanish, Portuguese, Japanese
Intended Use: The model is intended to be used in enterprise applications that involve processing of speech inputs. In particular, the model is well-suited for English, French, German, Spanish, Portuguese and Japanese speech-to-text and speech translations to and from English for the same languages, plus English-to-Italian and English-to-Mandarin.
Granite Speech model is supported natively in transformers>=4.52.1. Below is a simple example of how to use the granite-4.0-1b-speech model.
transformersFirst, make sure to install a recent version of transformers:
pip install transformers torchaudio soundfile
Then run the code:
import torch
import torchaudio
from huggingface_hub import hf_hub_download
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor
device = "cuda" if torch.cuda.is_available() else "cpu"
model_name = "ibm-granite/granite-4.0-1b-speech"
processor = AutoProcessor.from_pretrained(model_name)
tokenizer = processor.tokenizer
model = AutoModelForSpeechSeq2Seq.from_pretrained(
model_name, device_map=device, torch_dtype=torch.bfloat16
)
# Load audio
audio_path = hf_hub_download(repo_id=model_name, filename="multilingual_sample.wav")
wav, sr = torchaudio.load(audio_path, normalize=True)
assert wav.shape[0] == 1 and sr == 16000 # mono, 16kHz
# Create text prompt
user_prompt = "can you transcribe the speech into a written format?"
# Add "Keywords: , ..." at the end for keyword biasing
chat = [
{"role": "user", "content": user_prompt},
]
prompt = tokenizer.apply_chat_template(chat, tokenize=False, add_generation_prompt=True)
# Run the processor + model
model_inputs = processor(prompt, wav, device=device, return_tensors="pt").to(device)
model_outputs = model.generate(
**model_inputs, max_new_tokens=200, do_sample=False, num_beams=1
)
# Transformers includes the input IDs in the response
num_input_tokens = model_inputs["input_ids"].shape[-1]
new_tokens = model_outputs[0, num_input_tokens:].unsqueeze(0)
output_text = tokenizer.batch_decode(
new_tokens, add_special_tokens=False, skip_special_tokens=True
)
print(f"STT output = {output_text[0]}")
vLLMFirst, make sure to install vLLM:
pip install vllm
from transformers import AutoTokenizer
from vllm import LLM, SamplingParams
from vllm.assets.audio import AudioAsset
model_id = "ibm-granite/granite-4.0-1b-speech"
tokenizer = AutoTokenizer.from_pretrained(model_id)
def get_prompt(question: str, has_audio: bool):
"""Build the input prompt to send to vLLM."""
if has_audio:
question = f"{question}"
chat = [
{
"role": "user",
"content": question
}
]
return tokenizer.apply_chat_template(chat, tokenize=False)
model = LLM(
model=model_id,
max_model_len=2048, # This may be needed for lower resource devices.
limit_mm_per_prompt={"audio": 1},
)
question = "can you transcribe the speech into a written format?"
prompt_with_audio = get_prompt(
question=question,
has_audio=True,
)
audio = AudioAsset("mary_had_lamb").audio_and_sample_rate
inputs = {
"prompt": prompt_with_audio,
"multi_modal_data": {
"audio": audio,
}
}
outputs = model.generate(
inputs,
sampling_params=SamplingParams(Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.