Cataloged from handy-computer/parakeet-tdt-0.6b-v3-gguf
GGUF conversions of nvidia/parakeet-tdt-0.6b-v3 for use with transcribe.cpp.
Ported from upstream commit 6d590f7, pinned 2026-04-16. Validated against the NeMo reference at transcribe.cpp commit bf0d0b7 on 2026-04-18.
Offline multilingual speech-to-text covering 25 European languages. A 0.6B-parameter Conformer encoder with a TDT/RNNT transducer decoder. Takes a 16 kHz mono WAV and produces a transcript with optional token-level timestamps. Not a streaming model and does not translate.
| Quantization | Download | Size | WER (LibriSpeech test-clean) |
|---|---|---|---|
| F32 | parakeet-tdt-0.6b-v3-F32.gguf | 2.51 GB | 1.95% |
| F16 | parakeet-tdt-0.6b-v3-F16.gguf | 1.26 GB | 1.95% |
| Q8_0 | parakeet-tdt-0.6b-v3-Q8_0.gguf | 740 MB | 1.94% |
| Q6_K | parakeet-tdt-0.6b-v3-Q6_K.gguf | 627 MB | 1.93% |
| Q5_K_M | parakeet-tdt-0.6b-v3-Q5_K_M.gguf | 565 MB | 1.92% |
| Q4_K_M | parakeet-tdt-0.6b-v3-Q4_K_M.gguf | 502 MB |
| 1.98% |
WER measured on the full LibriSpeech test-clean split (2620 utterances) with greedy transducer decoding and no external LM. F32 reference baseline: 1.95%. NVIDIA's self-reported number on the same split is 1.93%.
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-tdt-0.6b-v3-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.
Inherited from the base model: CC-BY-4.0. See the upstream model card for full terms.
The section below is reproduced from nvidia/parakeet-tdt-0.6b-v3 at commit
6d590f7for offline reference. The upstream card is the authoritative source.
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.
parakeet-tdt-0.6b-v3's key features are built on the foundation of its predecessor, parakeet-tdt-0.6b-v2, and include:
For full details on the model architecture, training methodology, datasets, and evaluation results, check out the Technical Report.
GOVERNING TERMS: Use of this model is governed by the CC-BY-4.0 license.
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.
What is Nemotron? NVIDIA Developer Nemotron NVIDIA Riva Speech NeMo Documentation

Figure 1: ASR WER comparison across different models. This does not include Punctuation and Capitalisation errors.
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.
Global
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.
Huggingface 08/14/2025
Architecture Type:
FastConformer-TDT
Network Architecture:
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 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.
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.
import nemo.collections.asr as nemo_asr
asr_model = nemo_asr.models.ASRModel.from_pretrained(model_name="nvidia/parakeet-tdt-0.6b-v3")
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)
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']}")
#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)
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.
Runtime Engine(s):
Supported Hardware Microarchitecture Compatibility:
[Preferred/Supported] Operating System(s):
Hardware Specific Requirements:
Atleast 2GB RAM for model to load. The bigger the RAM, the larger audio input it supports.
Current version: parakeet-tdt-0.6b-v3. Previous versions can be accessed here.
This model was trained using the NeMo toolkit [5], following the strategies below:
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.
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:
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
Labeling Method by dataset
Properties:
For multilingual ASR performance evaluation:
For English ASR performance evaluation:
Data Collection Method by dataset
Labeling Method by dataset
Properties:
The tables below summarizes the WER (%) using a Transducer decoder with greedy decoding (without an external language model):
| Language | Fleurs | MLS | CoVoST |
|---|---|---|---|
| Average WER ↓ | 11.97% | 7.83% | 11.98% |
| bg | 12.64% | - | - |
| cs | 11.01% | - | - |
| da | 18.41% | - | - |
| de | 5.04% | - | 4.84% |
| el | 20.70% | - | - |
| en | 4.85% | - | 6.80% |
| es | 3.45% | 4.39% | 3.41% |
| et | 17.73% | - | 22.04% |
| fi | 13.21% | - | - |
| fr | 5.15% | 4.97% | 6.05% |
| hr | 12.46% | - | - |
| hu | 15.72% | - | - |
| it | 3.00% | 10.08% | 3.69% |
| lt | 20.35% | - | - |
| lv | 22.84% | - | 38.36% |
| mt | 20.46% | - | - |
| nl | 7.48% | 12.78% | 6.50% |
| pl | 7.31% | 7.28% | - |
| pt | 4.76% | 7.50% | 3.96% |
| ro | 12.44% | - | - |
| ru | 5.51% | - | 3.00% |
| sk | 8.82% | - | - |
| sl | 24.03% | - | 31.80% |
| sv | 15.08% | - | 20.16% |
| uk | 6.79% | - | 5.10% |
Note: WERs are calculated after removing Punctuation and Capitalization from reference and predicted text.
| Model | Avg WER | AMI | Earnings-22 | GigaSpeech | LS test-clean | LS test-other | SPGI Speech | TEDLIUM-v3 | VoxPopuli |
|---|---|---|---|---|---|---|---|---|---|
parakeet-tdt-0.6b-v3 | 6.34% | 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]
Performance across different Signal-to-Noise Ratios (SNR) using MUSAN music and noise samples [14]:
| SNR Level | Avg WER | AMI | Earnings | GigaSpeech | LS test-clean | LS test-other | SPGI | Tedlium | VoxPopuli | Relative Change |
|---|---|---|---|---|---|---|---|---|---|---|
| Clean | 6.34% | 11.31% | 11.42% | 9.59% | 1.93% | 3.59% | 3.97% | 2.75% | 6.14% | - |
| SNR 10 | 7.12% | 13.99% | 11.79% | 9.96% | 2.15% | 4.55% | 4.45% | 3.05% | 6.99% | -12.28% |
| SNR 5 | 8.23% | 17.59% | 13.01% | 10.69% | 2.62% | 6.05% | 5.23% | 3.33% | 7.31% | -29.81% |
| SNR 0 | 11.66% | 24.44% | 17.34% | 13.60% | 4.82% | 10.38% | 8.41% | 5.39% | 8.91% | -83.97% |
| SNR -5 | 19.88% | 34.91% | 26.92% | 21.41% | 12.21% | 19.98% | 16.96% | 11.36% | 15.30% | -213.64% |
[7] Youtube-Commons
[8] MOSEL: 950,000 Hours of Speech Data for Open-Source Speech Foundation Model Training on EU Languages
Engine:
Test Hardware:
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.
| Field | Response |
|---|---|
| Participation considerations from adversely impacted groups protected classes in model design and testing | None |
| Measures taken to mitigate against unwanted bias | None |
| 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. |
| 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/ |
| 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. |