Instructions to use moonbox-dev/Voxtral-Mini-3B-2507-MLX with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- MLX
How to use moonbox-dev/Voxtral-Mini-3B-2507-MLX with MLX:
# Download the model from the Hub pip install huggingface_hub[hf_xet] hf download moonbox-dev/Voxtral-Mini-3B-2507-MLX --local-dir Voxtral-Mini-3B-2507-MLX
- Notebooks
- Google Colab
- Kaggle
- Local Apps Settings
- LM Studio
- Atomic Chat
Voxtral Mini 3B MLX
MLX builds of Voxtral Mini 3B, made by Moonbox for Apple Silicon Macs. Each precision is a folder; keep a folder's files together. Only the parts listed below are converted: the rest of the pipeline comes from the original repository unless the section below says otherwise.
Files
The last column is how far each build is from the original PyTorch model on the same inputs (see the section below for what was measured).
| Precision | File | Size | sha256 | Checked against the original |
|---|---|---|---|---|
| bf16 | bf16/config.json |
1.5 kB | 228eec39354d |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/generation_config.json |
0.1 kB | cebeed3d4a16 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/model-00001-of-00002.safetensors |
5.36 GB | d830758e4223 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/model-00002-of-00002.safetensors |
3.99 GB | 09315657e9bb |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/model.safetensors.index.json |
71.1 kB | f682c18bfcd6 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/params.json |
0.7 kB | 4a37e19f2524 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/preprocessor_config.json |
0.4 kB | 86d67d926e17 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| bf16 | bf16/tekken.json |
14.9 MB | 4aaf3836c2a5 |
WER 1.15 % (original 1.15 %); transcripts identical to the original's |
| fp16 | fp16/config.json |
1.5 kB | 228eec39354d |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/generation_config.json |
0.1 kB | cebeed3d4a16 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/model-00001-of-00002.safetensors |
5.36 GB | 482968139d78 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/model-00002-of-00002.safetensors |
3.99 GB | 11277c832b17 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/model.safetensors.index.json |
71.1 kB | f682c18bfcd6 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/params.json |
0.7 kB | 4a37e19f2524 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/preprocessor_config.json |
0.4 kB | 86d67d926e17 |
WER 1.15 %; transcripts identical to the original's |
| fp16 | fp16/tekken.json |
14.9 MB | 4aaf3836c2a5 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/config.json |
1.7 kB | 3a5aecd2c4ec |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/generation_config.json |
0.1 kB | cebeed3d4a16 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/model-00001-of-00002.safetensors |
5.37 GB | cf91a1b58fa3 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/model-00002-of-00002.safetensors |
197.3 MB | 5a4e719ef2c4 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/model.safetensors.index.json |
0.1 MB | f50814443ebb |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/params.json |
0.7 kB | 4a37e19f2524 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/preprocessor_config.json |
0.4 kB | 86d67d926e17 |
WER 1.15 %; transcripts identical to the original's |
| 8bit | 8bit/tekken.json |
14.9 MB | 4aaf3836c2a5 |
WER 1.15 %; transcripts identical to the original's |
| 6bit | 6bit/config.json |
1.7 kB | d4bf1a368f37 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/generation_config.json |
0.1 kB | cebeed3d4a16 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/model.safetensors |
4.56 GB | 0283b107fe41 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/model.safetensors.index.json |
96.0 kB | 0da4be47bad2 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/params.json |
0.7 kB | 4a37e19f2524 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/preprocessor_config.json |
0.4 kB | 86d67d926e17 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 6bit | 6bit/tekken.json |
14.9 MB | 4aaf3836c2a5 |
WER 0.96 %; 0.2 % of the original's words read differently |
| 4bit | 4bit/config.json |
1.7 kB | a113d6595258 |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/generation_config.json |
0.1 kB | cebeed3d4a16 |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/model.safetensors |
3.55 GB | 4edb404f259e |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/model.safetensors.index.json |
96.0 kB | f848f5c8ba10 |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/params.json |
0.7 kB | 4a37e19f2524 |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/preprocessor_config.json |
0.4 kB | 86d67d926e17 |
WER 1.15 %; 0.4 % of the original's words read differently |
| 4bit | 4bit/tekken.json |
14.9 MB | 4aaf3836c2a5 |
WER 1.15 %; 0.4 % of the original's words read differently |
How to run
Voxtral Mini 3B (Mistral's speech model: transcription in 8 languages, plus audio question answering and
summarization on the Ministral 3B decoder) for mlx-audio. Each folder is a
complete mlx-audio model: model.safetensors (the Whisper-style audio encoder, the projector and the language model),
config.json, tekken.json (the tokenizer) and the processor config. Tested with mlx-audio==0.6.5 (Voxtral's
processor needs mistral-common[audio]>=1.11.5):
from mlx_audio.stt.utils import load_model
model = load_model("8bit") # a downloaded folder
print(model.generate("speech.wav", language="en", max_tokens=256).text)
8bit/, 6bit/ and 4bit/ quantize the language model and projector (MLX affine, group size 64); the audio encoder
stays bf16.
How they were checked. 24 LibriSpeech test-clean utterances in transcription mode, greedy decoding, with the
original in PyTorch (bfloat16, the checkpoint's dtype; transformers' VoxtralForConditionalGeneration) and every MLX
build: WER against the reference transcripts and against the original's own output. Pass: WER at most 1 point above
the original's. English only: the lab's other test language (Chinese) isn't one of the model's.
| Precision | Size | WER | Words read differently from the original | Real-time factor, M5 24 GB |
|---|---|---|---|---|
| original (PyTorch bf16, MPS) | - | 1.15 % | - | 0.28 |
bf16/ |
9.37 GB | 1.15 % | 0 | 0.29 |
fp16/ |
9.37 GB | 1.15 % | 0 | 0.29 |
8bit/ |
5.58 GB | 1.15 % | 0 | 0.19 |
6bit/ |
4.57 GB | 0.96 % | 0.2 % | 0.19 |
4bit/ |
3.56 GB | 1.15 % | 0.4 % | 0.16 |
Real-time factor = transcription time / audio length (lower is faster), one utterance at a time. bf16/ needs about
10.5 GB of memory; 8bit/ 6.7 GB.
How these were made
From Voxtral Mini 3B at revision 3060fe34b3: the original weights loaded in MLX and stored as bf16 or fp16, or
quantized with MLX's affine quantization (8bit, 6bit, 4bit, group size 64; layers MLX can't quantize, such as
convolutions and norms, stay bf16). Every build was checked against the original before upload; the sha256 column
above is what the files must hash to.
License
Apache-2.0. These files are a derivative of the original model and carry its license: it is copied in this repository, and it governs your use of these files. The original model card follows unchanged.
Original model card
Voxtral Mini 1.0 (3B) - 2507
Voxtral Mini is an enhancement of Ministral 3B, incorporating state-of-the-art audio input capabilities while retaining best-in-class text performance. It excels at speech transcription, translation and audio understanding.
Learn more about Voxtral in our blog post here and our research paper.
Key Features
Voxtral builds upon Ministral-3B with powerful audio understanding capabilities.
- Dedicated transcription mode: Voxtral can operate in a pure speech transcription mode to maximize performance. By default, Voxtral automatically predicts the source audio language and transcribes the text accordingly
- Long-form context: With a 32k token context length, Voxtral handles audios up to 30 minutes for transcription, or 40 minutes for understanding
- Built-in Q&A and summarization: Supports asking questions directly through audio. Analyze audio and generate structured summaries without the need for separate ASR and language models
- Natively multilingual: Automatic language detection and state-of-the-art performance in the world’s most widely used languages (English, Spanish, French, Portuguese, Hindi, German, Dutch, Italian)
- Function-calling straight from voice: Enables direct triggering of backend functions, workflows, or API calls based on spoken user intents
- Highly capable at text: Retains the text understanding capabilities of its language model backbone, Ministral-3B
Benchmark Results
Audio
Average word error rate (WER) over the FLEURS, Mozilla Common Voice and Multilingual LibriSpeech benchmarks:
Text
Usage
The model can be used with the following frameworks;
vllm (recommended): See hereTransformers🤗: See here
Notes:
temperature=0.2andtop_p=0.95for chat completion (e.g. Audio Understanding) andtemperature=0.0for transcription- Multiple audios per message and multiple user turns with audio are supported
- System prompts are not yet supported
vLLM (recommended)
We recommend using this model with vLLM.
Installation
Make sure to install vllm >= 0.10.0, we recommend using uv:
uv pip install -U "vllm[audio]" --system
Doing so should automatically install mistral_common >= 1.8.1.
To check:
python -c "import mistral_common; print(mistral_common.__version__)"
Offline
You can test that your vLLM setup works as expected by cloning the vLLM repo:
git clone https://github.com/vllm-project/vllm && cd vllm
and then running:
python examples/offline_inference/audio_language.py --num-audios 2 --model-type voxtral
Serve
We recommend that you use Voxtral-Small-24B-2507 in a server/client setting.
- Spin up a server:
vllm serve mistralai/Voxtral-Mini-3B-2507 --tokenizer_mode mistral --config_format mistral --load_format mistral
Note: Running Voxtral-Mini-3B-2507 on GPU requires ~9.5 GB of GPU RAM in bf16 or fp16.
- To ping the client you can use a simple Python snippet. See the following examples.
Audio Instruct
Leverage the audio capabilities of Voxtral-Mini-3B-2507 to chat.
Make sure that your client has mistral-common with audio installed:
pip install --upgrade mistral_common\[audio\]
Python snippet
from mistral_common.protocol.instruct.messages import TextChunk, AudioChunk, UserMessage, AssistantMessage, RawAudio
from mistral_common.audio import Audio
from huggingface_hub import hf_hub_download
from openai import OpenAI
# Modify OpenAI's API key and API base to use vLLM's API server.
openai_api_key = "EMPTY"
openai_api_base = "http://<your-server-host>:8000/v1"
client = OpenAI(
api_key=openai_api_key,
base_url=openai_api_base,
)
models = client.models.list()
model = models.data[0].id
obama_file = hf_hub_download("patrickvonplaten/audio_samples", "obama.mp3", repo_type="dataset")
bcn_file = hf_hub_download("patrickvonplaten/audio_samples", "bcn_weather.mp3", repo_type="dataset")
def file_to_chunk(file: str) -> AudioChunk:
audio = Audio.from_file(file, strict=False)
return AudioChunk.from_audio(audio)
text_chunk = TextChunk(text="Which speaker is more inspiring? Why? How are they different from each other?")
user_msg = UserMessage(content=[file_to_chunk(obama_file), file_to_chunk(bcn_file), text_chunk]).to_openai()
print(30 * "=" + "USER 1" + 30 * "=")
print(text_chunk.text)
print("\n\n")
response = client.chat.completions.create(
model=model,
messages=[user_msg],
temperature=0.2,
top_p=0.95,
)
content = response.choices[0].message.content
print(30 * "=" + "BOT 1" + 30 * "=")
print(content)
print("\n\n")
# The speaker who is more inspiring is the one who delivered the farewell address, as they express
# gratitude, optimism, and a strong commitment to the nation and its citizens. They emphasize the importance of
# self-government and active citizenship, encouraging everyone to participate in the democratic process. In contrast,
# the other speaker provides a factual update on the weather in Barcelona, which is less inspiring as it
# lacks the emotional and motivational content of the farewell address.
# **Differences:**
# - The farewell address speaker focuses on the values and responsibilities of citizenship, encouraging active participation in democracy.
# - The weather update speaker provides factual information about the temperature in Barcelona, without any emotional or motivational content.
messages = [
user_msg,
AssistantMessage(content=content).to_openai(),
UserMessage(content="Ok, now please summarize the content of the first audio.").to_openai()
]
print(30 * "=" + "USER 2" + 30 * "=")
print(messages[-1]["content"])
print("\n\n")
response = client.chat.completions.create(
model=model,
messages=messages,
temperature=0.2,
top_p=0.95,
)
content = response.choices[0].message.content
print(30 * "=" + "BOT 2" + 30 * "=")
print(content)
Transcription
Voxtral-Mini-3B-2507 has powerful transcription capabilities!
Make sure that your client has mistral-common with audio installed:
pip install --upgrade mistral_common\[audio\]
Python snippet
from mistral_common.protocol.transcription.request import TranscriptionRequest
from mistral_common.protocol.instruct.messages import RawAudio
from mistral_common.audio import Audio
from huggingface_hub import hf_hub_download
from openai import OpenAI
# Modify OpenAI's API key and API base to use vLLM's API server.
openai_api_key = "EMPTY"
openai_api_base = "http://<your-server-host>:8000/v1"
client = OpenAI(
api_key=openai_api_key,
base_url=openai_api_base,
)
models = client.models.list()
model = models.data[0].id
obama_file = hf_hub_download("patrickvonplaten/audio_samples", "obama.mp3", repo_type="dataset")
audio = Audio.from_file(obama_file, strict=False)
audio = RawAudio.from_audio(audio)
req = TranscriptionRequest(model=model, audio=audio, language="en", temperature=0.0).to_openai(exclude=("top_p", "seed"))
response = client.audio.transcriptions.create(**req)
print(response)
Transformers 🤗
Starting with transformers >= 4.54.0 and above, you can run Voxtral natively!
Install Transformers:
pip install -U transformers
Make sure to have mistral-common >= 1.8.1 installed with audio dependencies:
pip install --upgrade "mistral-common[audio]"
Audio Instruct
➡️ multi-audio + text instruction
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
conversation = [
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/mary_had_lamb.mp3",
},
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/winning_call.mp3",
},
{"type": "text", "text": "What sport and what nursery rhyme are referenced?"},
],
}
]
inputs = processor.apply_chat_template(conversation)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated response:")
print("=" * 80)
print(decoded_outputs[0])
print("=" * 80)
➡️ multi-turn
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
conversation = [
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/obama.mp3",
},
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3",
},
{"type": "text", "text": "Describe briefly what you can hear."},
],
},
{
"role": "assistant",
"content": "The audio begins with the speaker delivering a farewell address in Chicago, reflecting on his eight years as president and expressing gratitude to the American people. The audio then transitions to a weather report, stating that it was 35 degrees in Barcelona the previous day, but the temperature would drop to minus 20 degrees the following day.",
},
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/winning_call.mp3",
},
{"type": "text", "text": "Ok, now compare this new audio with the previous one."},
],
},
]
inputs = processor.apply_chat_template(conversation)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated response:")
print("=" * 80)
print(decoded_outputs[0])
print("=" * 80)
➡️ text only
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
conversation = [
{
"role": "user",
"content": [
{
"type": "text",
"text": "Why should AI models be open-sourced?",
},
],
}
]
inputs = processor.apply_chat_template(conversation)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated response:")
print("=" * 80)
print(decoded_outputs[0])
print("=" * 80)
➡️ audio only
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
conversation = [
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/winning_call.mp3",
},
],
}
]
inputs = processor.apply_chat_template(conversation)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated response:")
print("=" * 80)
print(decoded_outputs[0])
print("=" * 80)
➡️ batched inference
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
conversations = [
[
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/obama.mp3",
},
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3",
},
{
"type": "text",
"text": "Who's speaking in the speach and what city's weather is being discussed?",
},
],
}
],
[
{
"role": "user",
"content": [
{
"type": "audio",
"path": "https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/winning_call.mp3",
},
{"type": "text", "text": "What can you tell me about this audio?"},
],
}
],
]
inputs = processor.apply_chat_template(conversations)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated responses:")
print("=" * 80)
for decoded_output in decoded_outputs:
print(decoded_output)
print("=" * 80)
Transcription
➡️ transcribe
from transformers import VoxtralForConditionalGeneration, AutoProcessor
import torch
device = "cuda"
repo_id = "mistralai/Voxtral-Mini-3B-2507"
processor = AutoProcessor.from_pretrained(repo_id)
model = VoxtralForConditionalGeneration.from_pretrained(repo_id, torch_dtype=torch.bfloat16, device_map=device)
inputs = processor.apply_transcription_request(language="en", audio="https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/obama.mp3", model_id=repo_id)
inputs = inputs.to(device, dtype=torch.bfloat16)
outputs = model.generate(**inputs, max_new_tokens=500)
decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True)
print("\nGenerated responses:")
print("=" * 80)
for decoded_output in decoded_outputs:
print(decoded_output)
print("=" * 80)
Quantized
Model tree for moonbox-dev/Voxtral-Mini-3B-2507-MLX
Base model
mistralai/Voxtral-Mini-3B-2507
