Massively Multilingual Speech (MMS) - Finetuned LID
This checkpoint is a model fine-tuned for speech language identification (LID) and part of Facebook's Massive Multilingual Speech project. This checkpoint is based on the Wav2Vec2 architecture and classifies raw audio input to a probability distribution over 126 output classes (each class representing a language). The checkpoint consists of 1 billion parameters and has been fine-tuned from facebook/mms-1b on 126 languages.
Table Of Content
Example
This MMS checkpoint can be used with Transformers to identify the spoken language of an audio. It can recognize the following 126 languages.
Let's look at a simple example.
First, we install transformers and some other libraries
pip install torch accelerate torchaudio datasets
pip install --upgrade transformers
Note: In order to use MMS you need to have at least transformers >= 4.30 installed. If the 4.30 version
is not yet available on PyPI make sure to install transformers from
source:
pip install git+https://github.com/huggingface/transformers.git
Next, we load a couple of audio samples via datasets. Make sure that the audio data is sampled to 16000 kHz.
from datasets import load_dataset, Audio
# English
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "en", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
en_sample = next(iter(stream_data))["audio"]["array"]
# Arabic
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "ar", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
ar_sample = next(iter(stream_data))["audio"]["array"]
Next, we load the model and processor
from transformers import Wav2Vec2ForSequenceClassification, AutoFeatureExtractor
import torch
model_id = "facebook/mms-lid-126"
processor = AutoFeatureExtractor.from_pretrained(model_id)
model = Wav2Vec2ForSequenceClassification.from_pretrained(model_id)
Now we process the audio data, pass the processed audio data to the model to classify it into a language, just like we usually do for Wav2Vec2 audio classification models such as ehcalabres/wav2vec2-lg-xlsr-en-speech-emotion-recognition
# English
inputs = processor(en_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
lang_id = torch.argmax(outputs, dim=-1)[0].item()
detected_lang = model.config.id2label[lang_id]
# 'eng'
# Arabic
inputs = processor(ar_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
lang_id = torch.argmax(outputs, dim=-1)[0].item()
detected_lang = model.config.id2label[lang_id]
# 'ara'
To see all the supported languages of a checkpoint, you can print out the language ids as follows:
processor.id2label.values()
For more details, about the architecture please have a look at the official docs.
Supported Languages
This model supports 126 languages. Unclick the following to toogle all supported languages of this checkpoint in ISO 639-3 code. You can find more details about the languages and their ISO 649-3 codes in the MMS Language Coverage Overview.
- ara
- cmn
- eng
- spa
- fra
- mlg
- swe
- por
- vie
- ful
- sun
- asm
- ben
- zlm
- kor
- ind
- hin
- tuk
- urd
- aze
- slv
- mon
- hau
- tel
- swh
- bod
- rus
- tur
- heb
- mar
- som
- tgl
- tat
- tha
- cat
- ron
- mal
- bel
- pol
- yor
- nld
- bul
- hat
- afr
- isl
- amh
- tam
- hun
- hrv
- lit
- cym
- fas
- mkd
- ell
- bos
- deu
- sqi
- jav
- nob
- uzb
- snd
- lat
- nya
- grn
- mya
- orm
- lin
- hye
- yue
- pan
- jpn
- kaz
- npi
- kat
- guj
- kan
- tgk
- ukr
- ces
- lav
- bak
- khm
- fao
- glg
- ltz
- lao
- mlt
- sin
- sna
- ita
- srp
- mri
- nno
- pus
- eus
- ory
- lug
- bre
- luo
- slk
- fin
- dan
- yid
- est
- ceb
- war
- san
- kir
- oci
- wol
- haw
- kam
- umb
- xho
- epo
- zul
- ibo
- abk
- ckb
- nso
- gle
- kea
- ast
- sco
- glv
- ina
Model details
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Developed by: Vineel Pratap et al.
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Model type: Multi-Lingual Automatic Speech Recognition model
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Language(s): 126 languages, see supported languages
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License: CC-BY-NC 4.0 license
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Num parameters: 1 billion
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Audio sampling rate: 16,000 kHz
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Cite as:
@article{pratap2023mms, title={Scaling Speech Technology to 1,000+ Languages}, author={Vineel Pratap and Andros Tjandra and Bowen Shi and Paden Tomasello and Arun Babu and Sayani Kundu and Ali Elkahky and Zhaoheng Ni and Apoorv Vyas and Maryam Fazel-Zarandi and Alexei Baevski and Yossi Adi and Xiaohui Zhang and Wei-Ning Hsu and Alexis Conneau and Michael Auli}, journal={arXiv}, year={2023} }