Omnilingual ASR: Open-Source Multilingual Speech Recognition for 1600+ Languages

Omnilingual ASR is an open-source speech recognition system supporting over 1,600 languages — including hundreds never previously covered by any ASR technology. Designed for broad accessibility, it enables new languages to be added with just a few paired examples without requiring specialized expertise or large datasets. By combining scalable zero-shot learning with a flexible model family, Omnilingual ASR aims to make speech technology more inclusive and adaptable for communities and researchers worldwide.

Our 7B-LLM-ASR system achieves state-of-the-art performance across 1,600+ languages, with character error rates (CER) below 10 for 78% of those languages.

Documentation

Quick Start

Installation & Basic Usage - Setup and first transcription
Inference Pipeline - Comprehensive transcription guide with batch processing, language conditioning, and context examples
Supported Languages - View the complete list of 1600+ supported languages

Models & Architecture

Model Specifications - Available models, parameters, and memory requirements
Architecture Overview - Technical details on W2V, CTC, and LLM model families
Asset Management - Configuration system for models, tokenizers, and datasets

Training & Data Pipeline

Data Preparation - End-to-end guide for multilingual dataset preparation, HuggingFace integration, and parquet processing
Training Recipes - Pre-configured workflows for CTC and LLM model training

Installation

The models were developed using fairseq2, a research-focused sequence modeling toolkit. While we provide a reference inference pipeline that works across platforms, audio support requires libsndfile (Mac: brew install libsndfile; Windows may need an additional setup).

# using pip
pip install omnilingual-asr

# using uv
uv add omnilingual-asr

Inference

from omnilingual_asr.models.inference.pipeline import ASRInferencePipeline

pipeline = ASRInferencePipeline(model_card="omniASR_LLM_7B")

audio_files = ["/path/to/eng_audio1.flac", "/path/to/deu_audio2.wav"]
lang = ["eng_Latn", "deu_Latn"]
transcriptions = pipeline.transcribe(audio_files, lang=lang, batch_size=2)

More details on running specific models can be found in the src/omnilingual_asr/models/inference directory.

⚠️ Important: Currently only audio files shorter than 40 seconds are accepted for inference. We plan to add support for transcribing unlimited-length audio files shortly.

Supported Languages

To view the full list of 1600+ supported languages, you can access the language list programmatically:

from omnilingual_asr.models.wav2vec2_llama.lang_ids import supported_langs

# Print all supported languages
print(f"Total supported languages: {len(supported_langs)}")
print(supported_langs)

# Check if a specific language is supported
if "eng_Latn" in supported_langs:
    print("English (Latin script) is supported!")

Languages follow the format {language_code}_{script}, for example eng_Latn - English (Latin script), cmn_Hans - Mandarin Chinese (Simplified), ...

Using the HuggingFace Dataset 🤗

We provide a large-scale multilingual speech dataset on HuggingFace under CC-BY-4.0 License: facebook/omnilingual-asr-corpus. This dataset can be directly used with our inference pipeline for evaluation or testing:

pip install "omnilingual-asr[data]"

from datasets import load_dataset
from omnilingual_asr.models.inference.pipeline import ASRInferencePipeline

# Load dataset for a specific language (e.g., Ligurian)
omni_dataset = load_dataset("facebook/omnilingual-asr-corpus", "lij_Latn", split="train", streaming=True)
batch = next(omni_dataset.iter(5))

# Convert to pipeline input format
audio_data = [{"waveform": x["array"], "sample_rate": x["sampling_rate"]}
              for x in batch["audio"]]

# Run inference
pipeline = ASRInferencePipeline(model_card="omniASR_LLM_7B")
transcriptions = pipeline.transcribe(audio_data, batch_size=2)

# Display results
for i, (transcription, original_text) in enumerate(zip(transcriptions, batch["raw_text"]), 1):
    print(f"\n Sample {i}:")
    print(f"   Ground Truth: {original_text}")
    print(f"   Predicted:    {transcription}")

Model Architectures

Model Name	Features	Parameters	Download Size (FP32)	Inference VRAM¹	Real-Time Factor¹ (relative speed)²
`omniASR_W2V_300M`	SSL	317_390_592	1.2 GiB
`omniASR_W2V_1B`	SSL	965_514_752	3.6 GiB
`omniASR_W2V_3B`	SSL	3_064_124_672	12.0 GiB
`omniASR_W2V_7B`	SSL	6_488_487_168	25.0 GiB
`omniASR_CTC_300M`	ASR	325_494_996	1.3 GiB	~2 GiB	0.001 (96x)
`omniASR_CTC_1B`	ASR	975_065_300	3.7 GiB	~3 GiB	0.002 (48x)
`omniASR_CTC_3B`	ASR	3_080_423_636	12.0 GiB	~8 GiB	0.003 (32x)
`omniASR_CTC_7B`	ASR	6_504_786_132	25.0 GiB	~15 GiB	0.006 (16x)
`omniASR_LLM_300M`	ASR with optional language conditioning	1_627_603_584	6.1 GiB	~5 GiB	0.090 (~1x)
`omniASR_LLM_1B`	ASR with optional language conditioning	2_275_710_592	8.5 GiB	~6 GiB	0.091 (~1x)
`omniASR_LLM_3B`	ASR with optional language conditioning	4_376_679_040	17.0 GiB	~10 GiB	0.093 (~1x)
`omniASR_LLM_7B`	ASR with optional language conditioning	7_801_041_536	30.0 GiB	~17 GiB	0.092 (~1x)
`omniASR_LLM_7B_ZS`	Zero-Shot ASR	7_810_900_608	30.0 GiB	~20 GiB	0.194 (~0.5x)
`omniASR_tokenizer`	Tokenizer for most of architectures (except omniASR_LLM_7B)	-	100 KiB	-
`omniASR_tokenizer_v7`	Tokenizer for omniASR_LLM_7B model	-	100 KiB	-

¹ (batch=1, audio_len=30s, BF16, A100)

² Relative speed to omniASR_LLM_7B

Model Download & Storage

Automatic Download: Models are automatically downloaded on first use during training or inference
Storage Location: Models are saved to ~/.cache/fairseq2/assets/

Architecture Documentation

We provide a high-level model architecture overview in the model directory (src/omnilingual_asr/models), with individual configurations for each model family in the respective directories:

SSL Models: src/omnilingual_asr/models/wav2vec2_ssl
CTC Models: src/omnilingual_asr/models/wav2vec2_asr
LLM Models: src/omnilingual_asr/models/wav2vec2_llama

Training

To further finetune the released checkpoints on your own data, use our data preparation guide followed by the finetuning recipe guide.

License

Omnilingual ASR code and models are released under the Apache 2.0.

Citation

If you use the omnilingual ASR model suite in your research and wish to cite us, please use the following BibTeX entry (arxiv version will be added soon)!

@misc{omnilingualasr2025,
    title={{Omnilingual ASR}: Open-Source Multilingual Speech Recognition for 1600+ Languages},
    author={{Omnilingual ASR Team} and Keren, Gil and Kozhevnikov, Artyom and Meng, Yen and Ropers, Christophe and Setzler, Matthew and Wang, Skyler and Adebara, Ife and Auli, Michael and Balioglu, Can and Chan, Kevin and Cheng, Chierh and Chuang, Joe and Droof, Caley and Duppenthaler, Mark and Duquenne, Paul-Ambroise and Erben, Alexander and Gao, Cynthia and Mejia Gonzalez, Gabriel and Lyu, Kehan and Miglani, Sagar and Pratap, Vineel and Sadagopan, Kaushik Ram and Saleem, Safiyyah and Turkatenko, Arina and Ventayol-Boada, Albert and Yong, Zheng-Xin and Chung, Yu-An and Maillard, Jean and Moritz, Rashel and Mourachko, Alexandre and Williamson, Mary and Yates, Shireen},
    year={2025},
    url={https://ai.meta.com/research/publications/omnilingual-asr-open-source-multilingual-speech-recognition-for-1600-languages/},
}

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