/ Модель

Характеристики

Лицензия
CC BY 4.0
Базовая модель
parakeet-rnnt-1.1b
Библиотека
transcribe.cpp
Опубликована
10 мая 2026

parakeet-rnnt-1.1b-gguf

Распознавание речиАудио → текст4 профессии

Parakeet RNNT 1.1B — англоязычная модель автоматического распознавания речи, совместимая с transcribe.cpp и распространяемая в формате GGUF.

11 088скачиваний за 30 дней1лайквеса обновлены15 сен 2026

Бенчмарки

Как модель справилась с задачами пользователей ailist и что о ней публикуют открытые лидерборды

Публичные оценки

Из открытых лидербордов. Сравнивайте результаты одного теста при одинаковых условиях — отсутствие оценки не значит, что модель хуже.

В подключённых источниках результатов для этой модели пока нет.

Что умеет модель

Описание составлено по карточке модели и метаданным Hugging Face

Модель разработана совместно командами NVIDIA NeMo и Suno.ai, а данный репозиторий содержит её конвертированные и квантованные версии на основе nvidia/parakeet-rnnt-1.1b.

Модель предназначена для офлайн-преобразования речи в текст с использованием жадного RNN-T-декодирования без внешней языковой модели. Она принимает монофонический WAV-аудиосигнал с частотой 16 кГц и возвращает текст на английском языке в нижнем регистре без пунктуации. Поддерживаются токены временных меток, но потоковый режим, перевод и автоматическое определение языка не поддерживаются.

Архитектура включает энкодер FastConformer-XL примерно на 1,1 млрд параметров и декодер RNN-T. Доступны варианты F32, F16, Q8_0, Q6_K, Q5_K_M и Q4_K_M размером примерно от 825 МБ до 4,28 ГБ. На наборе LibriSpeech test-clean опубликованная WER составляет около 1,41–1,46% в зависимости от квантования; для Fleurs English указана WER 4,45% для Q8_0. Ключевые ограничения — поддержка только английского языка, отсутствие пунктуации, перевода и потоковой обработки, а также необходимость запуска через transcribe.cpp. Лицензия — CC-BY-4.0, унаследованная от базовой модели.

Для каких профессий подходит4

Разделы каталога профессий, в задачах которых используется эта модель

Все профессии

Карточка модели

README от автора на Hugging Face, как он его опубликовал

Открыть на Hugging Face

parakeet-rnnt-1.1b: transcribe.cpp GGUF

GGUF conversions of nvidia/parakeet-rnnt-1.1b for use with transcribe.cpp.

Ported from upstream commit a07b19e, pinned 2026-05-10. Validated against the NeMo reference at transcribe.cpp commit 42528dd on 2026-05-10.

Offline English speech-to-text with greedy RNN-T decoding. A 1.1B-parameter FastConformer-XL encoder with an RNN-T transducer decoder. Output is lowercase, no punctuation. Not a streaming model and does not translate.

Downloads

QuantizationDownloadSizeWER (LibriSpeech test-clean)
F32parakeet-rnnt-1.1b-F32.gguf4.28 GB1.45%
F16parakeet-rnnt-1.1b-F16.gguf2.15 GB1.45%
Q8_0parakeet-rnnt-1.1b-Q8_0.gguf1.27 GB1.46%
Q6_Kparakeet-rnnt-1.1b-Q6_K.gguf1.04 GB1.43%
Q5_K_Mparakeet-rnnt-1.1b-Q5_K_M.gguf936 MB1.43%
Q4_K_Mparakeet-rnnt-1.1b-Q4_K_M.gguf825 MB1.41%

WER on the full LibriSpeech test-clean split (2,620 utterances), batch size 1, timestamps none. Figures without a commit were published before provenance was recorded.

Greedy RNN-T decoding, no external LM. F32 reference baseline: 1.45%. NVIDIA's self-reported number on the same split is 1.46%.

Usage

Build transcribe.cpp from source:

git clone git@github.com:handy-computer/transcribe.cpp.git
cd transcribe.cpp
cmake -B build && cmake --build build

Run on a 16 kHz mono WAV:

build/bin/transcribe-cli \
  -m parakeet-rnnt-1.1b-Q8_0.gguf \
  input.wav

If your audio isn't already 16 kHz mono WAV, convert it first:

ffmpeg -i input.mp3 -ar 16000 -ac 1 output.wav

See the transcribe.cpp model page for performance numbers, numerical validation, and reproduction steps.

License

Inherited from the base model: CC-BY-4.0. See the upstream model card for full terms.


Original Model Card

The section below is reproduced from nvidia/parakeet-rnnt-1.1b at commit a07b19e for offline reference. The upstream card is the authoritative source.

Parakeet RNNT 1.1B (en)

Model architecture | Model size | Language

parakeet-rnnt-1.1b is an ASR model that transcribes speech in lower case English alphabet. This model is jointly developed by NVIDIA NeMo and Suno.ai teams. It is an XXL version of FastConformer Transducer [1] (around 1.1B parameters) model. See the model architecture section and NeMo documentation for complete architecture details.

Licence/Terms Of Use

License to use this model is covered by the CC-BY-4.0. By downloading the public and release version of the model, you accept the terms and conditions of the CC-BY-4.0 license.

Discover more from NVIDIA:

For documentation, deployment guides, enterprise-ready APIs, and the latest open models—including Nemotron and other cutting-edge speech, translation, and generative AI—visit the NVIDIA Developer Portal at developer.nvidia.com. Join the community to access tools, support, and resources to accelerate your development with NVIDIA’s NeMo, Riva, NIM, and foundation models.

Explore more from NVIDIA:

What is Nemotron? NVIDIA Developer Nemotron NVIDIA Riva Speech NeMo Documentation

NVIDIA NeMo: Training

To train, fine-tune or play with the model you will need to install NVIDIA NeMo. We recommend you install it after you've installed latest PyTorch version.

pip install nemo_toolkit['all']

How to Use this Model

The model is available for use in the NeMo toolkit [3], and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset.

Automatically instantiate the model

import nemo.collections.asr as nemo_asr
asr_model = nemo_asr.models.EncDecRNNTBPEModel.from_pretrained(model_name="nvidia/parakeet-rnnt-1.1b")

Transcribing using Python

First, let's get a sample

wget https://dldata-public.s3.us-east-2.amazonaws.com/2086-149220-0033.wav

Then simply do:

output = asr_model.transcribe(['2086-149220-0033.wav'])
print(output[0].text)

Transcribing many audio files

python [NEMO_GIT_FOLDER]/examples/asr/transcribe_speech.py 
 pretrained_name="nvidia/parakeet-rnnt-1.1b" 
 audio_dir="<DIRECTORY CONTAINING AUDIO FILES>"

Input

This model accepts 16000 Hz mono-channel audio (wav files) as input.

Output

This model provides transcribed speech as a string for a given audio sample.

Model Architecture

FastConformer [1] is an optimized version of the Conformer model with 8x depthwise-separable convolutional downsampling. The model is trained in a multitask setup with a Transducer decoder (RNNT) loss. You may find more information on the details of FastConformer here: Fast-Conformer Model.

Training

The NeMo toolkit [3] was used for training the models for over several hundred epochs. These model are trained with this example script and this base config.

The tokenizers for these models were built using the text transcripts of the train set with this script.

Datasets

The model was trained on 64K hours of English speech collected and prepared by NVIDIA NeMo and Suno teams.

The training dataset consists of private subset with 40K hours of English speech plus 24K hours from the following public datasets:

  • Librispeech 960 hours of English speech
  • Fisher Corpus
  • Switchboard-1 Dataset
  • WSJ-0 and WSJ-1
  • National Speech Corpus (Part 1, Part 6)
  • VCTK
  • VoxPopuli (EN)
  • Europarl-ASR (EN)
  • Multilingual Librispeech (MLS EN) - 2,000 hour subset
  • Mozilla Common Voice (v7.0)
  • People's Speech - 12,000 hour subset

Performance

The performance of Automatic Speech Recognition models is measuring using Word Error Rate. Since this dataset is trained on multiple domains and a much larger corpus, it will generally perform better at transcribing audio in general.

The following tables summarizes the performance of the available models in this collection with the Transducer decoder. Performances of the ASR models are reported in terms of Word Error Rate (WER%) with greedy decoding.

VersionTokenizerVocabulary SizeAMIEarnings-22Giga SpeechLS test-cleanSPGI SpeechTEDLIUM-v3Vox PopuliCommon Voice
1.22.0SentencePiece Unigram102417.1014.119.961.462.473.113.925.39

These are greedy WER numbers without external LM. More details on evaluation can be found at HuggingFace ASR Leaderboard

NVIDIA Riva: Deployment

NVIDIA Riva, is an accelerated speech AI SDK deployable on-prem, in all clouds, multi-cloud, hybrid, on edge, and embedded. Additionally, Riva provides:

  • World-class out-of-the-box accuracy for the most common languages with model checkpoints trained on proprietary data with hundreds of thousands of GPU-compute hours
  • Best in class accuracy with run-time word boosting (e.g., brand and product names) and customization of acoustic model, language model, and inverse text normalization
  • Streaming speech recognition, Kubernetes compatible scaling, and enterprise-grade support

Although this model isn’t supported yet by Riva, the list of supported models is here.
Check out Riva live demo.

References

[1] Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition

[2] Google Sentencepiece Tokenizer

[3] NVIDIA NeMo Toolkit

[4] Suno.ai

[5] HuggingFace ASR Leaderboard

Как запустить

На своей GPU в ailist. Цена — по текущим предложениям провайдеров

GPU в ailist

Универсальный рантайм · transcribe.cpp

Нельзя запустить

GGUF-веса поддерживаются только для текстовых LLM (llama.cpp).

Ревизия весов
a57f23c

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