---
license: apache-2.0
library_name: transformers
tags: []
---

# R1-AQA 

<!-- Provide a quick summary of what the model is/does. -->

## Introduction

R1-AQA extends `Qwen2-Audio-7B-Instruc` by integrating group relative policy optimization (GRPO). This adaptation enhances the model's capacity for temporal reasoning and contextual alignment in audio question answering (AQA) tasks.  
For more details, please refer to our [Github](https://github.com/xiaomi/r1-aqa) and [Report]().


## Inference
```python
import torch
import torchaudio
from transformers import Qwen2AudioForConditionalGeneration, AutoProcessor

# Load model
model_name = "mispeech/r1-aqa"
processor = AutoProcessor.from_pretrained(model_name)
model = Qwen2AudioForConditionalGeneration.from_pretrained(model_name, torch_dtype=torch.bfloat16, device_map="auto")

# Load example audio
wav_path = "test-mini-audios/3fe64f3d-282c-4bc8-a753-68f8f6c35652.wav"  # from MMAU dataset
waveform, sampling_rate = torchaudio.load(wav_path)
assert sampling_rate == 16000
audios = [waveform.numpy()]

# Make prompt text
question = "Based on the given audio, identify the source of the speaking voice."
options = ["Man", "Woman", "Child", "Robot"]
prompt = f"{question} Please choose the answer from the following options: {str(options)}. Output the final answer in <answer> </answer>."
message = [
    {"role": "user", "content": [
        {"type": "audio", "audio_url": wav_path},
        {"type": "text", "text": prompt}
    ]}
]
texts = processor.apply_chat_template(message, add_generation_prompt=True, tokenize=False)

# Process
inputs = processor(text=texts, audios=audios, sampling_rate=16000, return_tensors="pt", padding=True).to(model.device)
generated_ids = model.generate(**inputs, max_new_tokens=256)
generated_ids = generated_ids[:, inputs.input_ids.size(1):]
response = processor.batch_decode(generated_ids, skip_special_tokens=True, clean_up_tokenization_spaces=False)

print(response)
```