alt.freeweights.speech / parakeet-tdt-0.6b-v3
From: nvidia <huggingface.co/nvidia>Newsgroups: alt.freeweights.speechSubject:

parakeet-tdt-0.6b-v3

Date: 2026-11-15Commit: 541d1f9Signed: 2DC28CC7B26A283B ✓

Automatic speech recognition model with 627M parameters. Gguf and safetensors weights, 5.73 GB in all. Published by nvidia under CC-BY-4.0.

627M params5.73GBparakeet-tdt-0.6b-v3gguf safetensorspermissive
FactsParams627MSize5.73 GBFormatsgguf, safetensorsTaskautomatic speech recognitionLicenseCC-BY-4.0From sourcepermissive
$ fw get nvidia/parakeet-tdt-0.6b-v3
# manifest sha256 7e5b5788caaa… · 5.73 GB · 12 files
$ fw verify ./parakeet-tdt-0.6b-v3 nvidia/parakeet-tdt-0.6b-v3
From sourceHosted (signed)nvidia/parakeet-tdt-0.6b-v3Last check: 2026-11-15 02:26 UTC

Get help: custom fine-tune or private deploy. We review your setup before we quote.

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Overview

Size
5.73 GB
Weights
714 MB (GGUF)
Parameters
627M
Formats
gguf, safetensors
Task
automatic speech recognition
License
CC-BY-4.0 (permissive)
Files
12, each with a sha256
Publisher
nvidia
Added
2026-11-15
Section
alt.freeweights.speech
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Model card

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🦜 parakeet-tdt-0.6b-v3: Multilingual Speech-to-Text Model#

Model architecture | Model size | Language

Description:#

parakeet-tdt-0.6b-v3 is a 600-million-parameter multilingual automatic speech recognition (ASR) model designed for high-throughput speech-to-text transcription. It extends the parakeet-tdt-0.6b-v2 model by expanding language support from English to 25 European languages. The model automatically detects the language of the audio and transcribes it without requiring additional prompting. It is part of a series of models that leverage the Granary [1, 2] multilingual corpus as their primary training dataset.

🗣️ Try Demo here: https://huggingface.co/spaces/nvidia/parakeet-tdt-0.6b-v3

Supported Languages: Bulgarian (bg), Croatian (hr), Czech (cs), Danish (da), Dutch (nl), English (en), Estonian (et), Finnish (fi), French (fr), German (de), Greek (el), Hungarian (hu), Italian (it), Latvian (lv), Lithuanian (lt), Maltese (mt), Polish (pl), Portuguese (pt), Romanian (ro), Slovak (sk), Slovenian (sl), Spanish (es), Swedish (sv), Russian (ru), Ukrainian (uk)

This model is ready for commercial/non-commercial use.

Key Features:#

parakeet-tdt-0.6b-v3's key features are built on the foundation of its predecessor, parakeet-tdt-0.6b-v2, and include:

  • Automatic punctuation and capitalization
  • Accurate word-level and segment-level timestamps
  • Long audio transcription, supporting audio up to 24 minutes long with full attention (on A100 80GB) or up to 3 hours with local attention.
  • Released under a permissive CC BY 4.0 license

For full details on the model architecture, training methodology, datasets, and evaluation results, check out the Technical Report.

License/Terms of Use:#

GOVERNING TERMS: Use of this model is governed by 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

Automatic Speech Recognition (ASR) Performance#

Image: ASR WER Comparison

Figure 1: ASR WER comparison across different models. This does not include Punctuation and Capitalisation errors.


Evaluation Notes#

Note 1: The above evaluations are conducted for 24 supported languages, excluding Latvian since seamless-m4t-v2-large and seamless-m4t-medium do not support it.

Note 2: Performance differences may be partly attributed to Portuguese variant differences - our training data uses European Portuguese while most benchmarks use Brazilian Portuguese.

Deployment Geography:#

Global

Use Case:#

This model serves developers, researchers, academics, and industries building applications that require speech-to-text capabilities, including but not limited to: conversational AI, voice assistants, transcription services, subtitle generation, and voice analytics platforms.

Release Date:#

Huggingface 08/14/2025

Model Architecture:#

Architecture Type:

FastConformer-TDT

Network Architecture:

  • This model was developed based on FastConformer encoder architecture[3] and TDT decoder[4]
  • This model has 600 million model parameters.
Input:#

Input Type(s): 16kHz Audio Input Format(s): .wav and .flac audio formats Input Parameters: 1D (audio signal) Other Properties Related to Input: Monochannel audio

Output:#

Output Type(s): Text Output Format: String Output Parameters: 1D (text) Other Properties Related to Output: Punctuations and Capitalizations included.

Our AI models are designed and/or optimized to run on NVIDIA GPU-accelerated systems. By leveraging NVIDIA's hardware (e.g. GPU cores) and software frameworks (e.g., CUDA libraries), the model achieves faster training and inference times compared to CPU-only solutions.

For more information, refer to the NeMo documentation.

How to Use this Model:#

There are several ways to use this model. Choose the one that fits your needs.

Run locally with NeMo-Speech.cpp#

NeMo-Speech.cpp provides a lightweight native C++ runtime for local inference with this model. After installing the runtime:

hf download nvidia/parakeet-tdt-0.6b-v3 \
  parakeet-tdt-0.6b-v3.q8_0.gguf \
  --local-dir models

nemo-speech transcribe audio.wav \
  --model models/parakeet-tdt-0.6b-v3.q8_0.gguf

See the NeMo-Speech.cpp documentation for more details.

NVIDIA NeMo#

To train, fine-tune, or run Python inference with this model, install NVIDIA NeMo after installing a recent PyTorch version.

pip install -U nemo_toolkit['asr']

The model is available for use in the NeMo toolkit [5], and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset.

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

Automatically instantiate the model#
import nemo.collections.asr as nemo_asr
asr_model = nemo_asr.models.ASRModel.from_pretrained(model_name="nvidia/parakeet-tdt-0.6b-v3")
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 with timestamps#

To transcribe with timestamps:

output = asr_model.transcribe(['2086-149220-0033.wav'], timestamps=True)
# by default, timestamps are enabled for char, word and segment level
word_timestamps = output[0].timestamp['word'] # word level timestamps for first sample
segment_timestamps = output[0].timestamp['segment'] # segment level timestamps
char_timestamps = output[0].timestamp['char'] # char level timestamps

for stamp in segment_timestamps:
    print(f"{stamp['start']}s - {stamp['end']}s : {stamp['segment']}")
Transcribing long-form audio#
#updating self-attention model of fast-conformer encoder
#setting attention left and right context sizes to 256
asr_model.change_attention_model(self_attention_model="rel_pos_local_attn", att_context_size=[256, 256])

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

print(output[0].text)
Streaming with Parakeet models#

To use parakeet models in streaming mode use this script as shown below:

python NeMo/main/examples/asr/asr_chunked_inference/rnnt/speech_to_text_streaming_infer_rnnt.py \
    pretrained_name="nvidia/parakeet-tdt-0.6b-v3" \
    model_path=null \
    audio_dir="<optional path to folder of audio files>" \
    dataset_manifest="<optional path to manifest>" \
    output_filename="<optional output filename>" \
    right_context_secs=2.0 \
    chunk_secs=2 \
    left_context_secs=10.0 \
    batch_size=32 \
    clean_groundtruth_text=False

NVIDIA NIM for v2 parakeet model is available at https://build.nvidia.com/nvidia/parakeet-tdt-0_6b-v2.

Transformers 🤗 usage#

Until Parakeet TDT is part of an official Transformers release, you can use it by installing from source.

pip install git+https://github.com/huggingface/transformers

➡️ Pipeline usage

from transformers import pipeline

pipe = pipeline("automatic-speech-recognition", model="nvidia/parakeet-tdt-0.6b-v3")
out = pipe("https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3")
print(out)

➡️ AutoModel

from transformers import AutoModelForTDT, AutoProcessor
from datasets import load_dataset, Audio
import torch

device = "cuda" if torch.cuda.is_available() else "cpu"
num_samples = 3

model_id = "nvidia/parakeet-tdt-0.6b-v3"
processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForTDT.from_pretrained(model_id, dtype="auto", device_map=device)

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))

➡️ Timestamping

from datasets import Audio, load_dataset
from transformers import AutoModelForTDT, AutoProcessor
import torch

device = "cuda" if torch.cuda.is_available() else "cpu"
num_samples = 3

model_id = "nvidia/parakeet-tdt-0.6b-v3"
processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForTDT.from_pretrained(model_id, dtype="auto", device_map=device)

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)

➡️ Training

from transformers import AutoModelForTDT, AutoProcessor
from datasets import load_dataset, Audio
import torch

device = "cuda" if torch.cuda.is_available() else "cpu"

model_id = "nvidia/parakeet-tdt-0.6b-v3"
NUM_SAMPLES = 4

processor = AutoProcessor.from_pretrained(model_id)
model = AutoModelForTDT.from_pretrained(model_id, dtype=torch.bfloat16, device_map=device)
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_extractor.sampling_rate)
inputs.to(device=model.device, dtype=model.dtype)

outputs = model(**inputs)
print("Loss:", outputs.loss.item())
outputs.loss.backward()

For more details about usage, please refer to the Transformers' documentation.

Software Integration:#

Runtime Engine(s):

  • NeMo 2.4

Supported Hardware Microarchitecture Compatibility:

  • NVIDIA Ampere
  • NVIDIA Blackwell
  • NVIDIA Hopper
  • NVIDIA Volta

[Preferred/Supported] Operating System(s):

  • Linux

Hardware Specific Requirements:

At least 2GB RAM for model to load. The bigger the RAM, the larger audio input it supports.

Model Version#

Current version: parakeet-tdt-0.6b-v3. Previous versions can be accessed here.

Training and Evaluation Datasets:#

Training#

This model was trained using the NeMo toolkit [5], following the strategies below:

  • Initialized from a CTC multilingual checkpoint pretrained on the Granary dataset \[1] \[2].
  • Trained for 150,000 steps on 128 A100 GPUs.
  • Dataset corpora and languages were balanced using a temperature sampling value of 0.5.
  • Stage 2 fine-tuning was performed for 5,000 steps on 4 A100 GPUs using approximately 7,500 hours of high-quality, human-transcribed data of NeMo ASR Set 3.0.

Training was conducted using this example script and TDT configuration.

During the training, a unified SentencePiece Tokenizer \[6] with a vocabulary of 8,192 tokens was used. The unified tokenizer was constructed from the training set transcripts using this script and was optimized across all 25 supported languages.

Training Dataset#

The model was trained on the combination of Granary dataset's ASR subset and in-house dataset NeMo ASR Set 3.0:

  • 10,000 hours from human-transcribed NeMo ASR Set 3.0, including:
    • LibriSpeech (960 hours)
    • Fisher Corpus
    • National Speech Corpus Part 1
    • VCTK
    • Europarl-ASR
    • Multilingual LibriSpeech
    • Mozilla Common Voice (v7.0)
    • AMI
  • 660,000 hours of pseudo-labeled data from Granary \[1] \[2], including:

All transcriptions preserve punctuation and capitalization. The Granary dataset will be made publicly available after presentation at Interspeech 2025.

Data Collection Method by dataset

  • Hybrid: Automated, Human

Labeling Method by dataset

  • Hybrid: Synthetic, Human

Properties:

  • Noise robust data from various sources
  • Single channel, 16kHz sampled data
Evaluation Datasets#

For multilingual ASR performance evaluation:

  • Fleurs [10]
  • MLS [11]
  • CoVoST [12]

For English ASR performance evaluation:

  • Hugging Face Open ASR Leaderboard [13] datasets

Data Collection Method by dataset

  • Human

Labeling Method by dataset

  • Human

Properties:

  • All are commonly used for benchmarking English ASR systems.
  • Audio data is typically processed into a 16kHz mono channel format for ASR evaluation, consistent with benchmarks like the Open ASR Leaderboard.

Performance#

Multilingual ASR#

The tables below summarizes the WER (%) using a Transducer decoder with greedy decoding (without an external language model):

LanguageFleursMLSCoVoST
Average WER ↓11.97%7.83%11.98%
bg12.64%--
cs11.01%--
da18.41%--
de5.04%-4.84%
el20.70%--
en4.85%-6.80%
es3.45%4.39%3.41%
et17.73%-22.04%
fi13.21%--
fr5.15%4.97%6.05%
hr12.46%--
hu15.72%--
it3.00%10.08%3.69%
lt20.35%--
lv22.84%-38.36%
mt20.46%--
nl7.48%12.78%6.50%
pl7.31%7.28%-
pt4.76%7.50%3.96%
ro12.44%--
ru5.51%-3.00%
sk8.82%--
sl24.03%-31.80%
sv15.08%-20.16%
uk6.79%-5.10%

Note: WERs are calculated after removing Punctuation and Capitalization from reference and predicted text.

Huggingface Open-ASR-Leaderboard#
ModelAvg WERAMIEarnings-22GigaSpeechLS test-cleanLS test-otherSPGI SpeechTEDLIUM-v3VoxPopuli
parakeet-tdt-0.6b-v36.34%11.31%11.42%9.59%1.93%3.59%3.97%2.75%6.14%

Additional evaluation details are available on the Hugging Face ASR Leaderboard.[13]

Noise Robustness#

Performance across different Signal-to-Noise Ratios (SNR) using MUSAN music and noise samples [14]:

SNR LevelAvg WERAMIEarningsGigaSpeechLS test-cleanLS test-otherSPGITedliumVoxPopuliRelative Change
Clean6.34%11.31%11.42%9.59%1.93%3.59%3.97%2.75%6.14%-
SNR 107.12%13.99%11.79%9.96%2.15%4.55%4.45%3.05%6.99%-12.28%
SNR 58.23%17.59%13.01%10.69%2.62%6.05%5.23%3.33%7.31%-29.81%
SNR 011.66%24.44%17.34%13.60%4.82%10.38%8.41%5.39%8.91%-83.97%
SNR -519.88%34.91%26.92%21.41%12.21%19.98%16.96%11.36%15.30%-213.64%

References#

[1] Granary: Speech Recognition and Translation Dataset in 25 European Languages

[2] NVIDIA Granary Dataset Card

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

[4] Efficient Sequence Transduction by Jointly Predicting Tokens and Durations

[5] NVIDIA NeMo Toolkit

[6] Google Sentencepiece Tokenizer

[7] Youtube-Commons

[8] MOSEL: 950,000 Hours of Speech Data for Open-Source Speech Foundation Model Training on EU Languages

[9] YODAS: Youtube-Oriented Dataset for Audio and Speech

[10] FLEURS: Few-shot Learning Evaluation of Universal Representations of Speech

[11] MLS: A Large-Scale Multilingual Dataset for Speech Research

[12] CoVoST 2 and Massively Multilingual Speech-to-Text Translation

[13] HuggingFace ASR Leaderboard

[14] MUSAN: A Music, Speech, and Noise Corpus

Inference:#

Engine:

  • NVIDIA NeMo

Test Hardware:

  • NVIDIA A10
  • NVIDIA A100
  • NVIDIA A30
  • NVIDIA H100
  • NVIDIA L4
  • NVIDIA L40
  • NVIDIA Turing T4
  • NVIDIA Volta V100

Ethical Considerations:#

NVIDIA believes Trustworthy AI is a shared responsibility and we have established policies and practices to enable development for a wide array of AI applications. When downloaded or used in accordance with our terms of service, developers should work with their supporting model team to ensure this model meets requirements for the relevant industry and use case and addresses unforeseen product misuse.

For more detailed information on ethical considerations for this model, please see the Model Card++ Explainability, Bias, Safety & Security, and Privacy Subcards here.

Please report security vulnerabilities or NVIDIA AI Concerns here.

Bias:#

Field | Response ---------------------------------------------------------------------------------------------------|--------------- Participation considerations from adversely impacted groups protected classes in model design and testing | None Measures taken to mitigate against unwanted bias | None

Explainability:#

Field | Response ------------------------------------------------------------------------------------------------------|--------------------------------------------------------------------------------- Intended Domain | Speech to Text Transcription Model Type | FastConformer Intended Users | This model is intended for developers, researchers, academics, and industries building conversational based applications. Output | Text Describe how the model works | Speech input is encoded into embeddings and passed into conformer-based model and output a text response. Name the adversely impacted groups this has been tested to deliver comparable outcomes regardless of | Not Applicable Technical Limitations & Mitigation | Transcripts may be not 100% accurate. Accuracy varies based on language and characteristics of input audio (Domain, Use Case, Accent, Noise, Speech Type, Context of speech, etc.) Verified to have met prescribed NVIDIA quality standards | Yes Performance Metrics | Word Error Rate Potential Known Risks | If a word is not trained in the language model and not presented in vocabulary, the word is not likely to be recognized. Not recommended for word-for-word/incomplete sentences as accuracy varies based on the context of input text Licensing | GOVERNING TERMS: Use of this model is governed by the CC-BY-4.0 license.

Privacy:#

Field | Response ----------------------------------------------------------------------------------------------------------------------------------|----------------------------------------------- Generatable or reverse engineerable personal data? | None Personal data used to create this model? | None Is there provenance for all datasets used in training? | Yes Does data labeling (annotation, metadata) comply with privacy laws? | Yes Is data compliant with data subject requests for data correction or removal, if such a request was made? | No, not possible with externally-sourced data. Applicable Privacy Policy | https://www.nvidia.com/en-us/about-nvidia/privacy-policy/

Safety:#

Field | Response ---------------------------------------------------|---------------------------------- Model Application(s) | Speech to Text Transcription Describe the life critical impact | None Use Case Restrictions | Abide by CC-BY-4.0 License Model and dataset restrictions | The Principle of least privilege (PoLP) is applied limiting access for dataset generation and model development. Restrictions enforce dataset access during training, and dataset license constraints adhered to.

Files

12 files · 5.73 GBWeights 714 MB (GGUF). Sizes and sha256 from the signed manifest.
Hide all 12 files with their sha256
FileSizesha256
.eval_results/open_asr_leaderboard.yaml2.13 KB0fc9e33d2bef30eef17538cc5c5c3192bd65ff19207b82b1e21c6af893471bd9
.gitattributes1.72 KBa09613f18bdf229a26826f2d023431c0dd86f954531cb156ceeed1e2d18ca504
README.md42.9 KB7a4306a43f395e3a716e77f1d81c03db18a0a355d7b566bb1d6307213ba0302b
config.json1.15 KBe747b85e1bdfd300c8b8ac63bac8dd5221f8fe9bc275b48d06c735fcd6971b6e
generation_config.json289 Bb141de6ec6d7f982ece13f98f604e3fe1807ea9c0e839185d0ab7064604209d0
model.safetensors2.51 GB3a2026366188c8c68598edbbff92f8d11590a08e0ae2e6775544e7b07d6a5e11
parakeet-tdt-0.6b-v3.nemo2.51 GB3cbdc85877e668ca7b82d0d56770eb1fac76691f55d6b97545e8d61ca588d10d
parakeet-tdt-0.6b-v3.q8_0.gguf714 MBe3880d0aaaaf2c308ea2c35016b2b895c423eb3fda924c1b463d1c19b7f4d32e
plots/asr.png114 KB43df117825ce8148acce53c1c35cb54e9ecd111b835b171ff903b380429ee105
processor_config.json392 B8346a93a3b987fa1dec57a78f045cd0817d21786589a5a096b41a57a446fd1d7
tokenizer.json1.16 MBbd321b096832a3f270bd3b2a88823957920f1a5c5ada71114a26ea729d0cbe91
tokenizer_config.json290 B0b2fe0037599ee335f0b972fa682bf0ece74e4ccfec755cb7daa3405d3d3e874
VersionsEach time the pinned upstream commit or file list changed in this repo. Newest first.
Upstream commitDateContents
541d1f99c6 pinned now2026-11-15 02:26 UTC12 files · 5.73 GB

Use it

$ fw get nvidia/parakeet-tdt-0.6b-v3
# manifest sha256 7e5b5788caaa… · 5.73 GB · 12 files
$ fw verify ./parakeet-tdt-0.6b-v3 nvidia/parakeet-tdt-0.6b-v3
fw.pyChecks the signature and every sha256, tries each source in order, resumes.
$ python3 fw.py get https://freeweights.allternit.com/manifests/nvidia/parakeet-tdt-0.6b-v3.json --dir ./parakeet-tdt-0.6b-v3 $ python3 fw.py verify ./parakeet-tdt-0.6b-v3 ./nvidia__parakeet-tdt-0.6b-v3.json
hfThe Hugging Face CLI with one setting. Same repo id.
$ HF_ENDPOINT=https://freeweights.allternit.com hf download nvidia/parakeet-tdt-0.6b-v3 --revision 541d1f99c6b0c3cd0b11a95167540bb8edefd82b --local-dir ./parakeet-tdt-0.6b-v3
transformersSpeech recognition with the transformers pipeline.
import os os.environ["HF_ENDPOINT"] = "https://freeweights.allternit.com" from transformers import pipeline asr = pipeline("automatic-speech-recognition", model="nvidia/parakeet-tdt-0.6b-v3", revision="541d1f99c6b0c3cd0b11a95167540bb8edefd82b") print(asr("audio.wav")["text"])
llama.cppDownload one GGUF variant (parakeet-tdt-0.6b-v3.q8_0, 714 MB), then serve it.
$ HF_ENDPOINT=https://freeweights.allternit.com hf download nvidia/parakeet-tdt-0.6b-v3 --include "parakeet-tdt-0.6b-v3.q8_0*" --local-dir ./parakeet-tdt-0.6b-v3 $ llama-server -m ./parakeet-tdt-0.6b-v3/parakeet-tdt-0.6b-v3.q8_0.gguf
OllamaMake an Ollama model from the downloaded file.
$ printf 'FROM ./parakeet-tdt-0.6b-v3/parakeet-tdt-0.6b-v3.q8_0.gguf\n' > Modelfile $ ollama create parakeet-tdt-0.6b-v3 -f Modelfile

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Provenance

From: nvidia <huggingface.co/nvidia>Newsgroups: alt.freeweights.speechSubject: parakeet-tdt-0.6b-v3Date: 2026-11-15Commit: 541d1f9Signed: 2DC28CC7B26A283B ✓
Upstream
nvidia/parakeet-tdt-0.6b-v3
Pinned commit
541d1f99c6b0c3cd0b11a95167540bb8edefd82b
Manifest
parakeet-tdt-0.6b-v3.json · .minisig
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7e5b5788caaa734e5dcd5acf834b58eaf4eeaba6032b3d6fddda650846c6db20
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minisign, Ed25519, key 2DC28CC7B26A283B
Created
2026-11-15 02:26 UTC
Generator
freeweights-manifest/0.1.0
Takedowns
None.

How to verify

$ curl -O https://freeweights.allternit.com/manifests/nvidia/parakeet-tdt-0.6b-v3.json $ curl -O https://freeweights.allternit.com/manifests/nvidia/parakeet-tdt-0.6b-v3.json.minisig $ minisign -Vm parakeet-tdt-0.6b-v3.json -P RWQ7KGqyx4zCLXtUFOqvfv8L3g+8fpKJ1FsuxPyOYMF+4KufbUxKq1Oq $ shasum -a 256 <file> # compare with the Files tab

A manifest whose signature fails must be rejected. A file whose size or sha256 does not match must be discarded. Verify a download

Model tree

1 model in the catalog name this one as their parent.

Fine-tunes
1
Get it
Download
$ HF_ENDPOINT=https://freeweights.allternit.com hf download nvidia/parakeet-tdt-0.6b-v3
Open in Freeweights Studio

Pull, verify and run it in the Allternit app on Desktop, web or phone.

Signed manifest
minisign, key 2DC28CC7B26A283B. parakeet-tdt-0.6b-v3.json. Provenance. Takedown history: none.
From: MIT <huggingface.co/MIT>Subject: ast-finetuned-audioset-10-10-0.4593
From: onnx-community <huggingface.co/onnx-community>Subject: ast-finetuned-audioset-10-10-0.4593-ONNX
From: Edge0 <huggingface.co/Edge0>Subject: Audio8-TTS-Preview-0.6b
From: Edge0 <huggingface.co/Edge0>Subject: Audio8-TTS-Realistic-Air
From: facebook <huggingface.co/facebook>Subject: audiobox-aesthetics
From: mlx-community <huggingface.co/mlx-community>Subject: chatterbox-4bit