Model reference · open weights
granite-speech-3.3 is an open-weight audio or speech model from IBM. granite-speech-3.3-8b (BF16) weighs 17.4 GB; the smallest configuration that runs it is RTX 4090 24 GB.
Summary of the ibm-granite/granite-speech-3.3-2b model card, 2026-10-01. The estimate below is for another build of the family.
What it is
| Released by | IBM |
|---|---|
| Released | 2025-04-28 |
| Parameters | 3.0B |
| VRAM | 17.4 GB for the weights |
What it runs on
| Card | One stream | Memory |
|---|---|---|
| RTX 3060 12 GB … RTX 4060 Ti 16 GB | does not fit | |
| 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 |
Builds
| Build | Params | Precision | Weights | Smallest setup |
|---|---|---|---|---|
| granite-speech-3.3-2b ↗ | 3.0B | BF16 | 11.8 GB | RTX 4060 Ti 16 GB |
| granite-speech-3.3-8b (above) ↗ | 8.6B | BF16 | 17.4 GB | RTX 4090 24 GB |
From the model card
Model Summary: Granite-speech-3.3-2b is a compact and efficient speech-language model, specifically designed for automatic speech recognition (ASR) and automatic speech translation (AST). Granite-speech-3.3-2b uses a two-pass design, unlike integrated models that combine speech and language into a single pass. Initial calls to granite-speech-3.3-2b will transcribe audio files into text. To process the transcribed text using the underlying Granite language model, users must make a second call as each step must be explicitly initiated.
The model was trained on a collection of public corpora comprising diverse datasets for ASR and AST as well as synthetic datasets tailored to support the speech translation task. Granite-speech-3.3-2b was trained by modality aligning granite-3.3-2b-instruct (https://huggingface.co/ibm-granite/granite-3.3-2b-instruct) to speech on publicly available open source corpora containing audio inputs and text targets. Compared to the initial release, revision 3.3.2
Evaluations:
We evaluated granite-speech-3.3-2b revision 3.3.2 alongside granite-speech-3.3-8b (https://huggingface.co/ibm-granite/granite-speech-3.3-8b) and 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.
Release Date: June 19, 2025
License: Apache 2.0
Supported Languages: English, French, German, Spanish, Portuguese
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 and Portuguese speech-to-text and speech translations to and from English for the same languages plus English-to-Japanese and English-to-Mandarin. The model can also be used for tasks that involve text-only input since it calls the underlying granite-3.3-2b-instruct when the user specifies a prompt that does not contain audio.
Granite Speech model is supported natively in transformers from the main branch. Below is a simple example of how to use the granite-speech-3.3-2b revision 3.3.2 model.
transformersFirst, make sure to install a recent version of transformers:
pip install transformers>=4.52.4 torchaudio peft soundfile
Then run the code:
import torch
import torchaudio
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
from huggingface_hub import hf_hub_download
device = "cuda" if torch.cuda.is_available() else "cpu"
model_name = "ibm-granite/granite-speech-3.3-2b"
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="10226_10111_000000.wav")
wav, sr = torchaudio.load(audio_path, normalize=True)
assert wav.shape[0] == 1 and sr == 16000 # mono, 16khz
# create text prompt
system_prompt = "Knowledge Cutoff Date: April 2024.\nToday's Date: April 9, 2025.\nYou are Granite, developed by IBM. You are a helpful AI assistant"
user_prompt = "can you transcribe the speech into a written format?"
chat = [
dict(role="system", content=system_prompt),
dict(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 = torch.unsqueeze(model_outputs[0, num_input_tokens:], dim=0)
output_text = tokenizer.batch_decode(
new_tokens, add_special_tokens=False, skip_special_tokens=True
)
print(f"STT output = {output_text[0].upper()}")
vLLMFirst, make sure to install the latest version of vLLM:
pip install vllm --upgrade
from transformers import AutoTokenizer
from vllm import LLM, SamplingParams
from vllm.assets.audio import AudioAsset
from vllm.lora.request import LoRARequest
model_id = "ibm-granite/granite-speech-3.3-2b"
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)
# NOTE - you may see warnings about multimodal lora layers being ignored;
# this is okay as the lora in this model is only applied to the LLM.
model = LLM(
model=model_id,
enable_lora=True,
max_lora_rank=64,
max_model_len=2048, # This may be needed for lower resource devices.
limit_mm_per_prompt={"audio": 1},
)
### 1. Example with Audio [make sure to use the lora]
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(
temperature=0.2,
max_tokens=64,
),
lora_request=[LoRAReqQuoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.