This is a checkpoint for AL-Whisper-Instruct-R, audio-understanding reasoning language model, proposed in ALARM: Audio–Language Alignment for Reasoning Models. This model is trained only on the speech instruction subset of the data.

Can be combined with ALARM-CA to create ALARM-E model that achieves state-of-the-art performance (the best open-source, the third overall) on MMSU and MMAU-Speech benchmarks.

For more details regarding the model and its usage, please refer to our GitHub.

Inference

We provide vLLM support using vLLM Prompt Embedding API. Since ALARM uses the frozen Qwen3 model as the backbone, vllm just runs the original Qwen3 checkpoint, and the ALARM checkpoint is used for extracting LLM input embeddings. After you cloned the repo and installed the depnedencies, you can run the pretrained model as follows:

python

# Import libraries
import os
os.environ["CUDA_VISIBLE_DEVICES"] = "0" #optional

# run before importing torch because generate_vllm sets the multiprocessing method
from generate_vllm import get_response
from src.model.wrapped_llms.qwen3 import Qwen3AudioWrappedFeatureExtractor

from omegaconf import OmegaConf
from torchaudio.utils import _download_asset
from torchcodec.decoders import AudioDecoder
from transformers import AutoTokenizer
from vllm import LLM


# The model configuration config.
# Handles vllm-related configuration and defines feature extractors,
# i.e., audio -> encoder input embedding conversion.
# All other configuration, including model architecture, will be
# loaded from the checkpoint.
default_model_config_name = "src/configs/model/default_inference.yaml"
model_config = OmegaConf.load(default_model_config_name)

# checkpoint_name = which model to run
#   Single model version (no inference-time ensemble):
#   checkpoint_name='Blinorot/AL-Whisper-Instruct-R'
#   ALARM-E embedding fusion-type version (inference-time ensemble):
#   checkpoint_name=["Blinorot/ALARM-CA","Blinorot/AL-Whisper-Instruct-R"]
checkpoint_name = "Blinorot/AL-Whisper-Instruct-R"

device = "cuda"

# Load Tokenizer for Text Processing
tokenizer = AutoTokenizer.from_pretrained(model_config.llm)

# Load ALARM/AL-*-R checkpoints for extraction of LLM input embeddings
if isinstance(checkpoint_name, list): # ALARM-E-style embedding fusion (inference-time ensemble)
    feature_extractor_list = []
    for name in checkpoint_name:
        # Load weights into the (audio,text)->LLM embeddings converter
        feature_extractor = Qwen3AudioWrappedFeatureExtractor(
            model_config=model_config,
            checkpoint_name=name,
            tokenizer=tokenizer,
        )
        feature_extractor.to(device)
        feature_extractor_list.append(feature_extractor)
    feature_extractor = feature_extractor_list
else: # Single Model version (no inference-time ensemble)
    # Load weights into the (audio,text)->LLM embeddings converter
    feature_extractor = Qwen3AudioWrappedFeatureExtractor(
        model_config=model_config,
        checkpoint_name=checkpoint_name,
        tokenizer=tokenizer,
    )
    feature_extractor.to(device)

# Start the offline vLLM instance of original Qwen3 RLM
# Model will be loaded to CUDA_VISIBLE_DEVICES id
llm = LLM(
    model_config.llm,
    enable_prefix_caching=True,
    max_model_len=model_config.max_model_len,
    max_num_seqs=model_config.max_num_seq,
    max_num_batched_tokens=model_config.max_num_batched_tokens,
    gpu_memory_utilization=model_config.gpu_memory_utilization,
    enable_prompt_embeds=True,
)

# Set sampling arguments for the RLM
sample = llm.get_default_sampling_params()
sample.seed = model_config.seed
sample.max_tokens = model_config.max_tokens

# Define audio and prompt
# Audio must come from torchcodec.AudioDecoder
audio_example_path = _download_asset("tutorial-assets/ctc-decoding/1688-142285-0007.wav")
audio = AudioDecoder(audio_example_path)
prompt = "Describe the audio content."

# Define a system prompt
system_prompt = "You are an audio-understanding model."

# Obtain response from Audio RLM
response = get_response(
    prompts=[prompt], # list of all the prompts
    audio_list=[audio], # list of corresponding audio
    llm=llm,
    feature_extractor=feature_extractor,
    sample=sample,
    tokenizer=tokenizer,
    system_prompt=system_prompt,
    max_thinking_tokens=model_config.max_thinking_tokens, # controls thinking budget for the RLM
    debug=False,
)

# Response is a list of responses, one per each (prompt, audio) input pair
# We have only one input pair, so the final response is at index 0
response = response[0]

print(f"Model response:\n\n{response}")

Citation

If you use this work, please cite:

bibtex

@article{grinberg2026alarm,
  title={ALARM: Audio-Language Alignment for Reasoning Models},
  author={Grinberg, Petr and Shahmohammadi, Hassan},
  journal={arXiv preprint arXiv:2603.09556},
  year={2026}
}

License

The model checkpoint is licensed under Creative Commons Attribution-NonCommercial 4.0 (CC BY-NC 4.0). It may only be used for non-commercial research purposes.