Model reference · open weights

granite-4.0-speech

Audio ibm-granite Speech→text 1 build Open weights 40k dl/mo

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.

  • Granite-4.0-speech is a 2.3B parameter speech-language model developed by IBM for multilingual automatic speech recognition and bidirectional speech translation.
  • It supports English, French, German, Spanish, Portuguese, and Japanese, and is licensed under Apache 2.0.
  • The model features a 128k context length and is designed for enterprise applications involving speech-to-text processing.

Summary of the ibm-granite/granite-4.0-1b-speech model card, 2026-10-01

What it is

Released byIBM
Released2026-02-27
Parameters2.3B
VRAM4.6 GB for the weights

What it runs on

Memory and cards for granite-4.0-1b-speech (BF16)

4.6 GBweights, 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 IBM says about granite-4.0-speech

Read 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:

  • Supports multilingual speech inputs in English, French, German, Spanish, Portuguese and Japanese,
  • Provides higher transcription accuracy for English ASR and faster inference through better encoder training and speculative decoding,
  • Has half the number of parameters of granite-speech-3.3-2b for running on resource-constrained devices,
  • Adds keyword list biasing capability for enhanced name and acronym recognition

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:

modelAverage WERRTFxAMIEarnings22GigaspeechLS CleanLS OtherSPGISpeechTedliumVoxpopuli
ibm-granite/granite-4.0-1b-speech5.52280.028.448.4810.141.422.853.893.15.84

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.

Generation:

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.

Usage with transformers

First, 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]}")

Usage with vLLM

First, make sure to install vLLM:

pip install vllm
  • Code for offline mode:
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.

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