---
title: 'Voxtral Small: 24.3B Multimodal Transformer'
url: https://www.emergentmind.com/topics/voxtral-small
type: topic
---

# Voxtral Small: 24.3B Multimodal Transformer

Voxtral Small is a 24.3 billion-parameter multimodal Transformer model designed for integrated spoken audio and text understanding, featuring the ability to transcribe, translate, and answer open-domain queries about up to 40 minutes of continuous audio within a 32K-token context window. The architecture combines a state-of-the-art frozen Whisper large-v3 speech encoder, a two-layer audio-language adapter, and a Mistral Small 3.1 large language model decoder. Voxtral Small achieves state-of-the-art benchmarks across transcription, translation, and speech understanding, while maintaining highly competitive text-only performance. The model is released under the Apache 2.0 license, with public weights and source code.

## 1. Architecture and Parameterization

Voxtral Small follows a three-stage sequential pipeline:

- **Audio Encoder:** A frozen Whisper large-v3 module (640M parameters) encodes 128-bin log-Mel spectrograms sampled at 50 Hz. Input waveforms are divided into non-overlapping 30-second windows, with positional encodings reset per chunk. Each chunk yields $E^{(i)} \in \mathbb{R}^{T_c \times d_{model}}$ where $T_c = 1500$ frames.
- **Audio-Language Adapter:** A two-layer MLP (≈52M parameters) acts as both projector and temporal down-sampler. It applies $H = \mathrm{Gelu}(E W_1 + b_1) W_2 + b_2$ and selects every fourth frame along the time axis, reducing $T$ to $T' = T/4$, e.g., $30{,}000 \rightarrow 7{,}500$ for a 40-minute input.
- **Language Decoder:** An auto-regressive Transformer using the Mistral Small 3.1 24B backbone (≈22.9B parameters plus 670M in text token embeddings) handles multimodal inference and generation.

**Parameter Breakdown**

| Component                | Parameter Count     | Description                        |
|--------------------------|--------------------|-------------------------------------|
| Audio Encoder            | 640M               | Frozen Whisper large-v3             |
| Audio-Language Adapter   | 52M                | 2-layer MLP downsampler             |
| Text Embeddings          | 670M               | Token embedding layer               |
| Language Decoder         | 22.9B              | Mistral Small 3.1 backbone          |
| **Total**                | **24.3B**          |                                    |

The architecture is engineered to accommodate up to 40 minutes of audio and a few thousand text tokens within the 32K-model context window without significant performance loss [2507.13264].

## 2. Training Methodology and Objectives

### Pretraining
Pretraining leverages a large corpus of (audio, transcript) pairs from public and proprietary sources, segmented via VAD. Each audio chunk $A_n$ is paired with $T_n$ (transcript, either annotated or pseudo-labeled). Two patterns are mixed with equal probability:
- **Audio-to-Text Repetition (<repeat> mode):** Input is $[\texttt{<bos>}, A_n, \texttt{<repeat>}]$ with target $T_n$, enforcing local alignment.
- **Cross-Modal Continuation (<next> mode):** Input $[\texttt{<bos>}, A_1, T_2, \dots, A_{N-1}, \texttt{<next>}]$, target $T_N$, fostering discourse continuity.

The pretraining loss is:
$$
L_{pre} = \mathbb{E}_{\mathrm{pattern},(A,T)}\left[ -\sum_{t=1}^{|T|} \log P_\theta(T_t|\,\text{context}) \right]
$$
Pretraining initially restricts optimization to the adapter; Whisper and decoder are frozen. Text-only LLM pretraining (cross-entropy on text corpora) is interleaved to maintain strong unimodal performance.

### Supervised Finetuning
Synthetic and real tasks spanning long-form audio (up to 40 minutes) are generated, including QA, summarization, and translation, utilizing TTS conversion to augment audio-only instruction datasets and reduce TTS-only bias. The objective remains token-wise cross-entropy. A special <transcribe> mode triggers verbatim ASR.

### Preference Alignment (Online DPO)
Direct Preference Optimization (DPO) is applied in an online setting. Candidate responses $y^+, y^-$ are sampled (temperature $T=0.5$), scored by a reward model $R$ on ASR transcriptions, and the loss
$$
L_{DPO} = -\mathbb{E}_{(x, y^+, y^-)}\left[\log \sigma\left(\log \pi_\theta(y^+|x) - \log \pi_\theta(y^-|x)\right)\right]
$$
is optimized to sharpen output quality and reduce hallucinations.

## 3. Empirical Performance

Voxtral Small establishes new state-of-the-art results on a battery of open and closed speech and multimodal benchmarks:

**Speech Recognition (WER %)**
- LibriSpeech Test Clean: 1.53% (vs. Whisper v3 1.84%, GPT-4o mini 1.92%)
- LibriSpeech Test Other: 3.14% (vs. 3.66%, 4.70%)
- Mozilla CV (non-Arabic avg): ≈5.7% (vs. 8.2%, 11.1%)
- FLEURS (all languages avg): ≈5.1% (vs. 5.9%, 6.2%)

**Speech Translation (BLEU)**
- en→de: 47.0 (vs. 44.5 GPT-4o mini, 44.6 Gemini 2.5)
- en→fr: 57.3 (vs. 52.7, 53.9)
- de→en: 56.6 (vs. 51.8, 39.4)
- it→en: 46.8 (vs. 41.5, 31.8)

**Speech Understanding Accuracy (%)**
- Average: 77.4%
- Llama QA: 71.7% (vs. 74.3% GPT-4o mini, 66.3% Gemini 2.5)
- Openbook QA: 88.4% (vs. 83.7%, 94.7%)
- GSM8K: 89.7% (vs. 90.8%, 94.2%)
- In-house SU: 86.6% (vs. 80.0%, 88.6%)

Text-only performance matches the Mistral 3.1 backbone to within statistical noise, enabling drop-in multimodal and unimodal operation [2507.13264].

## 4. Extended Context Window and Long-Form Audio Processing

Processing up to 40 minutes of audio within a single inference pass requires:

- **Chunk-Wise Encoder Attention:** Each 30-second audio window is processed independently (complexity $O(N_c^2)$, $N_c=1,500$ frames), and chunk embeddings are concatenated.
- **4× Temporal Downsampling:** The adapter reduces token count by a factor of 4, achieving $T' = (D \cdot 50\,\text{Hz})/4 = D \cdot 12.5\,\text{Hz}$. For $D=2,400$ s (40 min), $T' \approx 30,000$, leaving nearly 2,000 text tokens—fitting in the 32K-token decoder window.

Total decoder memory scales as $O((T' + L_{text})^2)$. This enables 40-minute audio plus conversation with no significant ASR/understanding loss compared to non-downsampled input. *A plausible implication is that further downsampling (beyond 4×) would be detrimental, incurring >1% WER loss on non-English benchmarks.*

## 5. Practical Usage, Licensing, and Deployment

- **License:** Apache 2.0; no GPU-usage fees; weights and code hosted at https://huggingface.co/mistralai/Voxtral-Small-24B-2507.
- **Size/Memory:** 24.3B parameters (~48 GB fp16). 4-bit quantization reduces RAM to ≈12 GB, enabling single-GPU (A6000/RTX 4090) hosting.
- **Recommended Hardware:** ≥24 GB VRAM for quantized inference; ≥48 GB VRAM (two GPUs/model-parallel) for fp16. CPU-only possible but impractically slow.
- **PyTorch Inference Example:**

```python
from transformers import VoxtralProcessor, VoxtralForConditionalGeneration
import torch
processor = VoxtralProcessor.from_pretrained("mistralai/Voxtral-Small-24B-2507")
model = VoxtralForConditionalGeneration.from_pretrained("mistralai/Voxtral-Small-24B-2507",
                                                       device_map="auto",
                                                       load_in_4bit=True)
# Load a 40-min audio file @16 kHz
speech, sr = librosa.load("long_audio.mp3", sr=16000)
inputs = processor(speech, sampling_rate=sr, return_tensors="pt")
# Generate a summary or answer:
outputs = model.generate(**inputs, max_new_tokens=512)
print(processor.decode(outputs[0]))
```

## 6. Limitations and Prospects

Several constraints and future directions are noted:

- **Chunk Padding:** 30-second padding is retained to prevent language-specific ASR degradation (e.g., +0.5% WER for French if removed).
- **Downsampling Limit:** >4× downsampling leads to significant WER increases for non-English audio.
- **Preference Optimization Trade-Off:** Online DPO improves speech understanding (SU) benchmarks (+1.7% judge score) but slightly increases English short-form WER (6.31% → 6.50%); future schedules may mitigate this.
- **Open Directions:** Planned enhancements include cross-chunk attention (removing fixed 30s boundaries), larger contexts via sparse/routing attention, integration of video and document images, and lighter backbone variants (sub-10B parameters) for even broader local deployment applicability.

Voxtral Small is the first open-weights model capable of simultaneously processing 40-minute audio and engaging in fluent, factually rigorous conversation across speech and text modalities, providing an open, locally runnable platform for multimodal research and application [2507.13264].

Source: https://www.emergentmind.com/topics/voxtral-small