Model reference · open weights

asr-conformer-loquacious

Available as managed deployment Audio speechbrain Speech→text 1 variants 35 dl/mo

asr-conformer-loquacious is an open-weight audio or speech model from speechbrain. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.

Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.

What it is

Makerspeechbrain
TypeAudio & music
TaskSpeech→text
Runs withspeechbrain
Released2025-02-06
Popularity35 downloads / month
LicenceOpen weights

About

What asr-conformer-loquacious is

This model is a contribution of the Samsung AI Center-Cambridge.

This repository provides all the necessary tools to perform automatic speech recognition from an end-to-end system pretrained on LargeScaleASR (EN) within SpeechBrain. For a better experience, we encourage you to learn more about SpeechBrain. The performance of the model is the following:

#paramsvalidation WERtest WERLibriSpeech test-otherCommonVoice 18 testVoxpopuli testGPUs
480M6.87.54.612.06.98xV100 32GB

If you want to train your own model on this dataset, please refer to the SpeechBrain toolkit.

Pipeline description

This ASR system is composed of 2 different but linked blocks:

  • Tokenizer (unigram) that transforms words into subword units and trained with the train transcriptions of the LargeScaleASR dataset.
  • Acoustic model made of a conformer encoder and a joint decoder with CTC + transformer. Hence, the decoding also incorporates the CTC probabilities.

The system is trained with recordings sampled at 16kHz (single channel). The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling transcribe_file if needed.

Install SpeechBrain

First of all, please install SpeechBrain with the following command:

pip install speechbrain

Please notice that we encourage you to read our tutorials and learn more about SpeechBrain.

Transcribing your own audio files (in English)

from speechbrain.inference.ASR import EncoderDecoderASR

# For a full decoding with a large beam size (can be slow):
asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-conformer-largescaleasr", savedir="pretrained_models/asr-conformer-largescaleasr")

# For greedy decoding:
asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-conformer-largescaleasr", savedir="pretrained_models/asr-conformer-largescaleasr", overrides={"test_beam_size":"1"})

# For Attn. only decoding (faster):
asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-conformer-largescaleasr", savedir="pretrained_models/asr-conformer-largescaleasr", overrides={"scorer":None, "ctc_weight_decode":0.0})

# For even faster decoding
asr_model.transcribe_file("speechbrain/asr-conformer-largescaleasr/example.wav")

Inference on GPU

To perform inference on the GPU, add run_opts={"device":"cuda"} when calling the from_hparams method.

Parallel Inference on a Batch

About SpeechBrain

  • Website: https://speechbrain.github.io/
  • Code: https://github.com/speechbrain/speechbrain/
  • HuggingFace: https://huggingface.co/speechbrain/

Citing SpeechBrain

Please, cite SpeechBrain if you use it for your research or business.


@inproceedings{Loquacious,
  title     = {Loquacious Set: 25,000 Hours of Transcribed and Diverse English Speech Recognition Data for Research and Commercial Use},
  author    = {Titouan Parcollet and Yuan Tseng and Shucong Zhang and Rogier van Dalen},
  year      = {2025},
  booktitle = {Interspeech 2025},
}

@article{speechbrainV1,
  author  = {Mirco Ravanelli and Titouan Parcollet and Adel Moumen and Sylvain de Langen and Cem Subakan and Peter Plantinga and Yingzhi Wang and Pooneh Mousavi and Luca Della Libera and Artem Ploujnikov and Francesco Paissan and Davide Borra and Salah Zaiem and Zeyu Zhao and Shucong Zhang and Georgios Karakasidis and Sung-Lin Yeh and Pierre Champion and Aku Rouhe and Rudolf Braun and Florian Mai and Juan Zuluaga-Gomez and Seyed Mahed Mousavi and Andreas Nautsch and Ha Nguyen and Xuechen Liu and Sangeet Sagar and Jarod Duret and Salima Mdhaffar and Ga{{\"e}}lle Laperri{{\`e}}re and Mickael Rouvier and Renato De Mori and Yannick Est{{\`e}}ve},
  title   = {Open-Source Conversational AI with SpeechBrain 1.0},
  journal = {Journal of Machine Learning Research},
  year    = {2024},
  volume  = {25},
  number  = {333},
  pages   = {1--11},
  url     = {http://jmlr.org/papers/v25/24-0991.html}
}

From the published model card. Full card on the HuggingFace links in the sidebar.

How it works

How audio & music work

Audio or textinputAudio modelrecognise / synthesiseText or audiooutputSpeech-to-text turns audio into text; text-to-speech and music models turn text into audio.

Using it via the API

Call it like any OpenAI endpoint

Once AxForge deploys asr-conformer-loquacious for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (asr-conformer-loquacious below is illustrative; you get the exact model name on deployment.)

$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \
  -H "Authorization: Bearer $AXFORGE_API_KEY" \
  -F model="asr-conformer-loquacious" -F file=@audio.mp3

Create an account — your API key is available in the console. 5M tokens/month currently included with every new account at launch.

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