Model reference · open weights

parakeet-rnnt

Audio nvidia Speech→text 2 builds Open weights 70k dl/mo

parakeet-rnnt is an open-weight audio or speech model from NVIDIA. parakeet-rnnt-1.1b (FP32) weighs 2.1 GB; the smallest configuration that runs it is RTX 3060 12 GB.

For self-hosting, compare this build's requirements below with the GPU you plan to rent.

  • parakeet-rnnt is an automatic speech recognition model developed by NVIDIA and Suno.ai that transcribes English audio into lower case text.
  • The model features 617M parameters and is based on the FastConformer Transducer architecture.
  • It is available under the CC-BY-4.0 license.

Summary of the nvidia/parakeet-rnnt-0.6b model card, 2026-10-01. The estimate below is for another build of the family.

What it is

Released byNVIDIA
Released2023-12-28
Parameters617M
VRAM2.1 GB for the weights

What it runs on

Memory and cards for parakeet-rnnt-1.1b (FP32)

2.1 GBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

Builds

Sizes, precisions & builds

BuildParamsPrecisionWeightsSmallest setup
parakeet-rnnt-0.6b ↗ 617MFP32 1.2 GBRTX 3060 12 GB
parakeet-rnnt-1.1b (above) ↗ 1.1BFP32 2.1 GBRTX 3060 12 GB

From the model card

What NVIDIA says about parakeet-rnnt

Read the model card

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parakeet-rnnt-0.6b 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 XL version of FastConformer Transducer [1] (around 600M 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

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.

You can also run Parakeet RNNT with Transformers 🤗 (more below).

1) NeMo usage

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']
Automatically instantiate the model
import nemo.collections.asr as nemo_asr
asr_model = nemo_asr.models.EncDecRNNTBPEModel.from_pretrained(model_name="nvidia/parakeet-rnnt-0.6b")
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-0.6b"
 audio_dir=""

2) Transformers 🤗 usage

Parakeet RNNT is available in 🤗 Transformers starting from v5.13.0.

pip install "transformers>=5.13.0"
from transformers import pipeline

pipe = pipeline("automatic-speech-recognition", model="nvidia/parakeet-rnnt-0.6b")
out = pipe("https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3")
print(out)
from transformers import AutoModelForRNNT, AutoProcessor
from datasets import load_dataset, Audio

num_samples = 3

model_id = "nvidia/parakeet-rnnt-0.6b"
processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForRNNT.from_pretrained(model_id, dtype="auto", device_map="auto")

ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
ds = ds.cast_column("audio", Audio(sampling_rate=processor.feature_extractor.sampling_rate))
speech_samples = [el["array"] for el in ds["audio"][:num_samples]]

inputs = processor(speech_samples, sampling_rate=processor.feature_extractor.sampling_rate)
inputs.to(model.device, dtype=model.dtype)
output = model.generate(**inputs, return_dict_in_generate=True)
print(processor.decode(output.sequences, skip_special_tokens=True))
from datasets import Audio, load_dataset
from transformers import AutoModelForRNNT, AutoProcessor

num_samples = 3

model_id = "nvidia/parakeet-rnnt-0.6b"
processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForRNNT.from_pretrained(model_id, dtype="auto", device_map="auto")

ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
ds = ds.cast_column("audio", Audio(sampling_rate=processor.feature_extractor.sampling_rate))
speech_samples = [el["array"] for el in ds["audio"][:num_samples]]

inputs = processor(speech_samples, sampling_rate=processor.feature_extractor.sampling_rate)
inputs.to(model.device, dtype=model.dtype)
output = model.generate(**inputs, return_dict_in_generate=True)
decoded_output, decoded_timestamps = processor.decode(
    output.sequences,
    durations=output.durations,
    skip_special_tokens=True,
)
print("Transcription:", decoded_output)
print("Timestamped tokens:", decoded_timestamps)
from transformers import AutoModelForRNNT, AutoProcessor
from datasets import load_dataset, Audio
import torch

model_id = "nvidia/parakeet-rnnt-0.6b"
NUM_SAMPLES = 4

processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForRNNT.from_pretrained(model_id, dtype=torch.bfloat16, device_map="auto")
model.train()

ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
ds = ds.cast_column("audio", Audio(sampling_rate=processor.feature_extractor.sampling_rate))
speech_samples = [el["array"] for el in ds["audio"][:NUM_SAMPLES]]
text_samples = ds["text"][:NUM_SAMPLES]

# passing `text` to the processor will prepare inputs' `labels` key
inputs = processor(audio=speech_samples, text=text_samples, sampling_rate=processor.feature_

Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.

Benchmarks

Reported results

As published on the model card: the maker's own numbers, not measured by AxForge.

TaskDatasetMetricScore
Automatic Speech RecognitionAMI (Meetings test)Test WER17.550
Automatic Speech RecognitionEarnings-22Test WER14.780
Automatic Speech RecognitionGigaSpeechTest WER10.070
Automatic Speech RecognitionLibriSpeech (clean)Test WER1.630
Automatic Speech RecognitionLibriSpeech (other)Test WER3.060
automatic-speech-recognitionSPGI SpeechTest WER3.470
automatic-speech-recognitiontedlium-v3Test WER3.860
Automatic Speech RecognitionVox PopuliTest WER6.050
automatic-speech-recognitionMozilla Common Voice 9.0Test WER8.070
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