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    Copyright © 2026 NVIDIA Corporation

    NVIDIA

    parakeet-tdt-0.6b-v2

    Downloadable

    Accurate and optimized English transcriptions with punctuation and word timestamps

    • ASR
    • English
    • NVIDIA NIM
    • NVIDIA Riva
    • speech-to-text
    Get API Key
    API ReferenceAPI Reference
    Accelerated by DGX Cloud

    This NIM contains 2 profiles:

    • nvidia/parakeet-tdt-0.6b-v2 (English only)
    • nvidia/parakeet-tdt-0.6b-v3 (25 European languages)

    Parakeet-tdt-0.6b-v2 is a 600-million-parameter automatic speech recognition (ASR) model designed for high-quality English transcription, featuring support for punctuation, capitalization, and accurate timestamp prediction.

    This XL variant of the FastConformer architecture integrates the TDT decoder and is trained with full attention, enabling efficient transcription of audio segments up to 24 minutes in a single pass.

    Key Features:

    • Accurate word-level timestamp predictions
    • Automatic punctuation and capitalization
    • Robust performance on spoken numbers, and song lyrics transcription

    GOVERNING TERMS: Use of this Parakeet-tdt-0.6b-v2 is governed by the NVIDIA Community Model License Agreement (found at NVIDIA Agreements | Enterprise Software | NVIDIA Community Model License).

    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.

    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.

    Try the experience 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)

    GOVERNING TERMS: Use of this Parakeet-tdt-0.6b-v3 is governed by the NVIDIA open model licence agreement (found at NVIDIA Open Model License Agreement).

    Automatic Speech Recognition (ASR) Performance

    ASR WER Comparison - click to view full size

    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:

    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 -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.

    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)
    

    Software Integration:

    Runtime Engine(s):

    • NeMo 2.2

    Supported Hardware Microarchitecture Compatibility:

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

    [Preferred/Supported] Operating System(s):

    • Linux

    Hardware Specific Requirements:

    Atleast 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:

      • YTC [7]
      • MOSEL [8]
      • YODAS [9]

    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.

    On this page

    1. Automatic Speech Recognition (ASR) Performance
      1. Evaluation Notes
      2. Deployment Geography
      3. Use Case
      4. Release Date
      5. Model Architecture
      6. Input
      7. Output
    2. How to Use this Model
    3. Software Integration
    4. Training and Evaluation Datasets
      1. Training
      2. Training Dataset
    5. Performance
      1. Noise Robustness
    6. References
    7. Inference
    8. Ethical Considerations