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*This model was published in HF papers on 2025-07-10 and contributed to Hugging Face Transformers on 2025-11-12.*

# Audio Flamingo 3

<div class="flex flex-wrap space-x-1">
<img alt="FlashAttention" src="https://img.shields.io/badge/%E2%9A%A1%EF%B8%8E%20FlashAttention-eae0c8?style=flat">
<img alt="SDPA" src="https://img.shields.io/badge/SDPA-DE3412?style=flat&logo=pytorch&logoColor=white">
</div>

## Overview

Audio Flamingo 3 (AF3) is a fully open large audio–language model designed for robust understanding and reasoning over speech, environmental sounds, and music. AF3 pairs a Whisper-style audio encoder with a causal language model and performs replace-in-place audio–text fusion: the processor aligns post-pool audio frames to a dedicated placeholder token and the model replaces those token slots with projected audio embeddings during the forward pass.

The model checkpoint is available at: [nvidia/audio-flamingo-3-hf](https://huggingface.co/nvidia/audio-flamingo-3-hf)

Highlights:

- Unified audio encoder across speech, sound, and music.
- **Long-audio support via windowing and post-pool alignment (up to 10 minutes maximum).** The model processes audio in 30-second windows with a hard limit of 20 windows (10 minutes total). Audio longer than 10 minutes will be truncated.
- Deterministic fusion that preserves sequence length by replacing audio placeholder tokens with audio embeddings.

This model was contributed by [Lasha Koroshinadze](https://huggingface.co/lashahub) and [Eric Bezzam](https://huggingface.co/bezzam).

### Paper

[Audio Flamingo 3](https://huggingface.co/papers/2507.08128): Advancing Audio Intelligence with Fully Open Large Audio Language Models  
A. Goel, S. Ghosh, J. Kim, S. Kumar, Z. Kong, S. Lee, C.-H. H. Yang, R. Duraiswami, D. Manocha, R. Valle, B. Catanzaro  
NVIDIA and University of Maryland  
Project: https://research.nvidia.com/labs/adlr/AF3/

## Usage

### Audio Instruct Mode

The model supports audio-text instructions, including multi-turn interactions, all processed in batches.

➡️ audio + text instruction

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

conversation = [
    {
        "role": "user",
        "content": [
            {"type": "text", "text": "Transcribe the input speech."},
            {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/WhDJDIviAOg_120_10.mp3"},
        ],
    }
]

inputs = processor.apply_chat_template(
    conversation,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
).to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)

decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print(decoded_outputs)
```

➡️ multi-turn:

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

conversation = [
    {
        "role": "user",
        "content": [
            {
                "type": "text",
                "text": "Instruction: How does the tone of female speech change throughout the audio? Choose the correct option among the options below: (A) Sad to happy (B) Happy to sad (C) Neutral to happy (D) Happy to neutral.",
            },
            {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/000000786159.31.wav"},
        ],
    },
    {
        "role": "assistant",
        "content": [{"type": "text", "text": "(A) Sad to happy"}],
    },
    {
        "role": "user",
        "content": [
            {"type": "text", "text": "Why do you think so?"},
        ],
    },
]

inputs = processor.apply_chat_template(
    conversation,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
).to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)

decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print(decoded_outputs)
```

➡️ text only:

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

conversation = [
    {
        "role": "user",
        "content": [
            {"type": "text", "text": "What is the capital of France?"},
        ],
    }
]

inputs = processor.apply_chat_template(
    conversation,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
).to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)

decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print(decoded_outputs)
```

➡️ audio only:

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

conversation = [
    {
        "role": "user",
        "content": [
            {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/WhDJDIviAOg_120_10.mp3"},
        ],
    }
]

inputs = processor.apply_chat_template(
    conversation,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
).to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)

decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print(decoded_outputs)
```

➡️ batched inference!

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

conversations = [
    [
        {
            "role": "user",
            "content": [
                {"type": "text", "text": "Transcribe the input speech."},
                {
                    "type": "audio",
                    "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/t_837b89f2-26aa-4ee2-bdf6-f73f0dd59b26.wav",
                },
            ],
        }
    ],
    [
        {
            "role": "user",
            "content": [
                {
                    "type": "text",
                    "text": "This track feels really peaceful and introspective. What elements make it feel so calming and meditative?",
                },
                {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/FPSbCAANfbJLVSwD.mp3"},
            ],
        }
    ],
]

inputs = processor.apply_chat_template(
    conversations,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
).to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)

decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print(decoded_outputs)
```

➡️ Training:

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")
model.train()

conversation = [
    [
        {
            "role": "user",
            "content": [
                {"type": "text", "text": "Transcribe the input speech."},
                {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/WhDJDIviAOg_120_10.mp3"},
            ],
        },
        {
            "role": "assistant",
            "content": [{"type": "text", "text": "The transcription of the audio is 'summer follows spring the days grow longer and the nights are warm'."}],
        }
    ],
    [
        {
            "role": "user",
            "content": [
                {
                    "type": "text",
                    "text": "This track feels really peaceful and introspective. What elements make it feel so calming and meditative?",
                },
                {"type": "audio", "path": "https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/FPSbCAANfbJLVSwD.mp3"},
            ],
        },
        {
            "role": "assistant",
            "content": [{"type": "text", "text": "The transcription of the audio is 'some transcription of the audio'."}],
        }

    ]
]

inputs = processor.apply_chat_template(
    conversation,
    tokenize=True,
    add_generation_prompt=True,
    return_dict=True,
    output_labels=True,
).to(model.device, dtype=model.dtype)

loss = model(**inputs).loss
loss.backward()
```

➡️ transcription shortcut

```python
from transformers import AudioFlamingo3ForConditionalGeneration, AutoProcessor


model_id = "nvidia/audio-flamingo-3-hf"
processor = AutoProcessor.from_pretrained(model_id)
model = AudioFlamingo3ForConditionalGeneration.from_pretrained(model_id, device_map="auto")

inputs = processor.apply_transcription_request(audio="https://huggingface.co/datasets/nvidia/AudioSkills/resolve/main/assets/t_837b89f2-26aa-4ee2-bdf6-f73f0dd59b26.wav").to(model.device, dtype=model.dtype)

outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True, strip_prefix=True)

print(decoded_outputs)
```

The model is trained to emit transcriptions prefixed with assistant framing such as `The spoken content of the audio is "<text>".`. Use `strip_prefix=True` (as shown above) to remove the fixed assistant sentence and surrounding quotes so that only the transcription remains.

## How the model works

### Architecture

* **AudioFlamingo3Encoder**
  Whisper-style feature extractor + encoder → average-pool over time (stride 2) → LayerNorm.
  Produces per-frame hidden states at the post-pool rate.

* **AudioFlamingo3MultiModalProjector**
  A small MLP that maps encoder features to the language model’s hidden size.

* **AudioFlamingo3ForConditionalGeneration**
  A causal language model that accepts text embeddings where each audio placeholder token slot is replaced, in place, by an audio frame embedding. No sequence-length change is introduced by fusion.

### Processor-level alignment

1. Each raw waveform is split into fixed-length windows based on the feature extractor’s `chunk_length` (seconds) and `sampling_rate` (Hz).
2. For each window, the processor computes the number of post-pool frames `post_pool_len` that the encoder will output (matching the conv/pool schedule).
3. The processor expands the audio placeholder token by the total number of post-pool frames across all windows.
4. The model later replaces those token positions with the corresponding projected audio embeddings.

## Usage patterns

### Transcription shortcut

For automatic speech recognition you can skip writing the default instruction each time and call
[`~transformers.AudioFlamingo3Processor.apply_transcription_request`]:

```python
inputs = processor.apply_transcription_request(audio=audio_array)
```

Pass `prompt="Transcribe the input speech."` (or a list of prompts for batch audio) to customize the instruction while
keeping the audio placeholder handling.

`audio` accepts in-memory arrays, local file paths, or URLs. Any processor kwargs (`text_kwargs`, `audio_kwargs`, etc.)
are forwarded, so you can tweak padding or tensor formats just like when calling `processor(...)`.

## Long audio and windowing

**Important: Maximum audio length is 10 minutes.** Audio longer than this will be truncated.

* The default setup processes 30-second windows at 16 kHz mono.
* **The processor enforces a hard limit of 20 windows per sample, resulting in a maximum of 10 minutes of audio (20 windows × 30 seconds).**
* For each window:

  * `mel_len` is the padded mel length.
  * A conv stack reduces time as `conv_output_len = (mel_len - 1) // 2 + 1`.
  * Post-pool frames per window: `post_pool_len = (conv_output_len - 2) // 2 + 1`.
  * An audio placeholder token is expanded to the sum of `post_pool_len` across all windows.

## Padding, attention, and caching

* **Left padding vs right padding**
  For generation with mixed prompt lengths in a batch, left padding is usually preferable.
  For training, right padding is common; AF3’s fusion mechanism itself is padding-agnostic because it replaces in place.
* **Attention masks**
  The processor returns `attention_mask` (text) and `input_features_mask` (audio). The model builds an internal 4-D mask on the encoder’s pre-pool axis with negative infinity at pad positions.
* **Caching**
  During generation, `input_features` and `input_features_mask` are only passed on the first step. Subsequent steps use cached keys/values from the language model.

## Troubleshooting

* Empty or truncated outputs when batching
  Use left padding for batched generation and decode only the new tokens after the prompt length, as shown in the quickstart.

## AudioFlamingo3Config

[[autodoc]] AudioFlamingo3Config

## AudioFlamingo3EncoderConfig

[[autodoc]] AudioFlamingo3EncoderConfig

## AudioFlamingo3Processor

[[autodoc]] AudioFlamingo3Processor
    - __call__

## AudioFlamingo3Encoder

[[autodoc]] AudioFlamingo3Encoder
    - forward

## AudioFlamingo3Model

[[autodoc]] AudioFlamingo3Model
    - forward

## AudioFlamingo3ForConditionalGeneration

[[autodoc]] AudioFlamingo3ForConditionalGeneration
    - forward
    - get_audio_features
