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- README.md +289 -0
- parakeet-ctc-0.6b-F16.gguf +3 -0
- parakeet-ctc-0.6b-F32.gguf +3 -0
- parakeet-ctc-0.6b-Q4_K_M.gguf +3 -0
- parakeet-ctc-0.6b-Q5_K_M.gguf +3 -0
- parakeet-ctc-0.6b-Q6_K.gguf +3 -0
- parakeet-ctc-0.6b-Q8_0.gguf +3 -0
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| 1 |
+
---
|
| 2 |
+
license: cc-by-4.0
|
| 3 |
+
base_model: nvidia/parakeet-ctc-0.6b
|
| 4 |
+
base_model_relation: quantized
|
| 5 |
+
library_name: transcribe.cpp
|
| 6 |
+
pipeline_tag: automatic-speech-recognition
|
| 7 |
+
language:
|
| 8 |
+
- en
|
| 9 |
+
tags:
|
| 10 |
+
- gguf
|
| 11 |
+
- transcribe.cpp
|
| 12 |
+
- asr
|
| 13 |
+
- speech-to-text
|
| 14 |
+
- parakeet
|
| 15 |
+
- conformer
|
| 16 |
+
- ctc
|
| 17 |
+
---
|
| 18 |
+
|
| 19 |
+
# parakeet-ctc-0.6b — transcribe.cpp GGUF
|
| 20 |
+
|
| 21 |
+
GGUF conversions of [nvidia/parakeet-ctc-0.6b](https://huggingface.co/nvidia/parakeet-ctc-0.6b) for use
|
| 22 |
+
with [transcribe.cpp](https://github.com/handy-computer/transcribe.cpp).
|
| 23 |
+
|
| 24 |
+
Ported from upstream commit
|
| 25 |
+
[ad09ba1](https://huggingface.co/nvidia/parakeet-ctc-0.6b/commit/ad09ba1),
|
| 26 |
+
pinned 2026-05-10.
|
| 27 |
+
Validated against the NeMo reference at transcribe.cpp commit
|
| 28 |
+
[2f7a6d2](https://github.com/handy-computer/transcribe.cpp/tree/2f7a6d2)
|
| 29 |
+
on 2026-05-10.
|
| 30 |
+
|
| 31 |
+
Offline English speech-to-text with greedy CTC decoding. A 0.6B-parameter FastConformer-Large encoder with a linear CTC head — the simplest and fastest decoder in the parakeet family. Output is lowercase, no punctuation. Not a streaming model and does not translate.
|
| 32 |
+
|
| 33 |
+
|
| 34 |
+
## Downloads
|
| 35 |
+
|
| 36 |
+
| Quantization | Download | Size | WER (LibriSpeech test-clean) |
|
| 37 |
+
| --- | --- | ---: | ---: |
|
| 38 |
+
| F32 | [parakeet-ctc-0.6b-F32.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-F32.gguf) | 2.44 GB | 1.87% |
|
| 39 |
+
| F16 | [parakeet-ctc-0.6b-F16.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-F16.gguf) | 1.22 GB | 1.87% |
|
| 40 |
+
| Q8_0 | [parakeet-ctc-0.6b-Q8_0.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-Q8_0.gguf) | 722 MB | 1.87% |
|
| 41 |
+
| Q6_K | [parakeet-ctc-0.6b-Q6_K.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-Q6_K.gguf) | 594 MB | 1.84% |
|
| 42 |
+
| Q5_K_M | [parakeet-ctc-0.6b-Q5_K_M.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-Q5_K_M.gguf) | 533 MB | 1.87% |
|
| 43 |
+
| Q4_K_M | [parakeet-ctc-0.6b-Q4_K_M.gguf](https://huggingface.co/handy-computer/parakeet-ctc-0.6b-gguf/resolve/main/parakeet-ctc-0.6b-Q4_K_M.gguf) | 469 MB | 1.90% |
|
| 44 |
+
|
| 45 |
+
WER measured on the full LibriSpeech test-clean split (2620 utterances) with greedy CTC decoding and no external LM. F32 reference baseline: 1.87%. NVIDIA's self-reported number on the same split is 1.87%.
|
| 46 |
+
|
| 47 |
+
|
| 48 |
+
## Usage
|
| 49 |
+
|
| 50 |
+
Build transcribe.cpp from source:
|
| 51 |
+
|
| 52 |
+
```bash
|
| 53 |
+
git clone git@github.com:handy-computer/transcribe.cpp.git
|
| 54 |
+
cd transcribe.cpp
|
| 55 |
+
cmake -B build && cmake --build build
|
| 56 |
+
```
|
| 57 |
+
|
| 58 |
+
Run on a 16 kHz mono WAV:
|
| 59 |
+
|
| 60 |
+
```bash
|
| 61 |
+
build/bin/transcribe-cli \
|
| 62 |
+
-m parakeet-ctc-0.6b-Q8_0.gguf \
|
| 63 |
+
input.wav
|
| 64 |
+
```
|
| 65 |
+
|
| 66 |
+
If your audio isn't already 16 kHz mono WAV, convert it first:
|
| 67 |
+
|
| 68 |
+
```bash
|
| 69 |
+
ffmpeg -i input.mp3 -ar 16000 -ac 1 output.wav
|
| 70 |
+
```
|
| 71 |
+
|
| 72 |
+
See the [transcribe.cpp model page](https://github.com/handy-computer/transcribe.cpp/blob/main/docs/models/parakeet-ctc-0.6b.md) for performance
|
| 73 |
+
numbers, numerical validation, and reproduction steps.
|
| 74 |
+
|
| 75 |
+
## License
|
| 76 |
+
|
| 77 |
+
Inherited from the base model: **CC-BY-4.0**. See the
|
| 78 |
+
[upstream model card](https://huggingface.co/nvidia/parakeet-ctc-0.6b) for full terms.
|
| 79 |
+
|
| 80 |
+
---
|
| 81 |
+
|
| 82 |
+
## Original Model Card
|
| 83 |
+
|
| 84 |
+
> The section below is reproduced from
|
| 85 |
+
> [nvidia/parakeet-ctc-0.6b](https://huggingface.co/nvidia/parakeet-ctc-0.6b) at commit
|
| 86 |
+
> `ad09ba1` for offline reference. The upstream card is the
|
| 87 |
+
> authoritative source.
|
| 88 |
+
|
| 89 |
+
# Parakeet CTC 0.6B (en)
|
| 90 |
+
|
| 91 |
+
<style>
|
| 92 |
+
img {
|
| 93 |
+
display: inline;
|
| 94 |
+
}
|
| 95 |
+
</style>
|
| 96 |
+
|
| 97 |
+
[](#model-architecture)
|
| 98 |
+
| [](#model-architecture)
|
| 99 |
+
| [](#datasets)
|
| 100 |
+
|
| 101 |
+
|
| 102 |
+
`parakeet-ctc-0.6b` is an ASR model that transcribes speech in lower case English alphabet. This model is jointly developed by [NVIDIA NeMo](https://github.com/NVIDIA/NeMo) and [Suno.ai](https://www.suno.ai/) teams.
|
| 103 |
+
It is an XL version of FastConformer CTC [1] (around 600M parameters) model.
|
| 104 |
+
See the [model architecture](#model-architecture) section and [NeMo documentation](https://docs.nvidia.com/deeplearning/nemo/user-guide/docs/en/main/asr/models.html#fast-conformer) for complete architecture details.
|
| 105 |
+
|
| 106 |
+
## NVIDIA NeMo: Training
|
| 107 |
+
|
| 108 |
+
To train, fine-tune or play with the model you will need to install [NVIDIA NeMo](https://github.com/NVIDIA/NeMo). We recommend you install it after you've installed latest PyTorch version.
|
| 109 |
+
```
|
| 110 |
+
pip install nemo_toolkit['all']
|
| 111 |
+
```
|
| 112 |
+
|
| 113 |
+
## How to Use this Model
|
| 114 |
+
|
| 115 |
+
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. Moreover, you can now run Parakeet CTC natively with [Transformers](https://github.com/huggingface/transformers) 🤗.
|
| 116 |
+
|
| 117 |
+
### Automatically instantiate the model
|
| 118 |
+
|
| 119 |
+
```python
|
| 120 |
+
import nemo.collections.asr as nemo_asr
|
| 121 |
+
asr_model = nemo_asr.models.EncDecCTCModelBPE.from_pretrained(model_name="nvidia/parakeet-ctc-0.6b")
|
| 122 |
+
```
|
| 123 |
+
|
| 124 |
+
### Transcribing using NeMo
|
| 125 |
+
First, let's get a sample
|
| 126 |
+
```
|
| 127 |
+
wget https://dldata-public.s3.us-east-2.amazonaws.com/2086-149220-0033.wav
|
| 128 |
+
```
|
| 129 |
+
Then simply do:
|
| 130 |
+
```
|
| 131 |
+
asr_model.transcribe(['2086-149220-0033.wav'])
|
| 132 |
+
```
|
| 133 |
+
|
| 134 |
+
### Transcribing using [Transformers](https://github.com/huggingface/transformers) 🤗
|
| 135 |
+
|
| 136 |
+
Make sure to install `transformers` from source.
|
| 137 |
+
|
| 138 |
+
```bash
|
| 139 |
+
pip install git+https://github.com/huggingface/transformers
|
| 140 |
+
```
|
| 141 |
+
|
| 142 |
+
<details>
|
| 143 |
+
<summary>➡️ Pipeline usage</summary>
|
| 144 |
+
|
| 145 |
+
```python
|
| 146 |
+
from transformers import pipeline
|
| 147 |
+
|
| 148 |
+
pipe = pipeline("automatic-speech-recognition", model="nvidia/parakeet-ctc-0.6b")
|
| 149 |
+
out = pipe("https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3")
|
| 150 |
+
print(out)
|
| 151 |
+
```
|
| 152 |
+
</details>
|
| 153 |
+
|
| 154 |
+
<details>
|
| 155 |
+
<summary>➡️ AutoModel</summary>
|
| 156 |
+
|
| 157 |
+
```python
|
| 158 |
+
from transformers import AutoModelForCTC, AutoProcessor
|
| 159 |
+
from datasets import load_dataset, Audio
|
| 160 |
+
import torch
|
| 161 |
+
|
| 162 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 163 |
+
|
| 164 |
+
processor = AutoProcessor.from_pretrained("nvidia/parakeet-ctc-0.6b")
|
| 165 |
+
model = AutoModelForCTC.from_pretrained("nvidia/parakeet-ctc-0.6b", dtype="auto", device_map=device)
|
| 166 |
+
|
| 167 |
+
ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
| 168 |
+
ds = ds.cast_column("audio", Audio(sampling_rate=processor.feature_extractor.sampling_rate))
|
| 169 |
+
speech_samples = [el['array'] for el in ds["audio"][:5]]
|
| 170 |
+
|
| 171 |
+
inputs = processor(speech_samples, sampling_rate=processor.feature_extractor.sampling_rate)
|
| 172 |
+
inputs.to(model.device, dtype=model.dtype)
|
| 173 |
+
outputs = model.generate(**inputs)
|
| 174 |
+
print(processor.batch_decode(outputs))
|
| 175 |
+
```
|
| 176 |
+
</details>
|
| 177 |
+
|
| 178 |
+
<details>
|
| 179 |
+
<summary>➡️ Training</summary>
|
| 180 |
+
|
| 181 |
+
```python
|
| 182 |
+
from transformers import AutoModelForCTC, AutoProcessor
|
| 183 |
+
from datasets import load_dataset, Audio
|
| 184 |
+
import torch
|
| 185 |
+
|
| 186 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 187 |
+
|
| 188 |
+
processor = AutoProcessor.from_pretrained("nvidia/parakeet-ctc-0.6b")
|
| 189 |
+
model = AutoModelForCTC.from_pretrained("nvidia/parakeet-ctc-0.6b", dtype="auto", device_map=device)
|
| 190 |
+
|
| 191 |
+
ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
|
| 192 |
+
ds = ds.cast_column("audio", Audio(sampling_rate=processor.feature_extractor.sampling_rate))
|
| 193 |
+
speech_samples = [el['array'] for el in ds["audio"][:5]]
|
| 194 |
+
text_samples = [el for el in ds["text"][:5]]
|
| 195 |
+
|
| 196 |
+
# passing `text` to the processor will prepare inputs' `labels` key
|
| 197 |
+
inputs = processor(audio=speech_samples, text=text_samples, sampling_rate=processor.feature_extractor.sampling_rate)
|
| 198 |
+
inputs.to(device, dtype=model.dtype)
|
| 199 |
+
|
| 200 |
+
outputs = model(**inputs)
|
| 201 |
+
outputs.loss.backward()
|
| 202 |
+
```
|
| 203 |
+
</details>
|
| 204 |
+
|
| 205 |
+
For more details about usage, the refer to [Transformers' documentation](https://huggingface.co/docs/transformers/en/index).
|
| 206 |
+
|
| 207 |
+
### Transcribing many audio files
|
| 208 |
+
|
| 209 |
+
```shell
|
| 210 |
+
python [NEMO_GIT_FOLDER]/examples/asr/transcribe_speech.py
|
| 211 |
+
pretrained_name="nvidia/parakeet-ctc-0.6b"
|
| 212 |
+
audio_dir="<DIRECTORY CONTAINING AUDIO FILES>"
|
| 213 |
+
```
|
| 214 |
+
|
| 215 |
+
### Input
|
| 216 |
+
|
| 217 |
+
This model accepts 16000 Hz mono-channel audio (wav files) as input.
|
| 218 |
+
|
| 219 |
+
### Output
|
| 220 |
+
|
| 221 |
+
This model provides transcribed speech as a string for a given audio sample.
|
| 222 |
+
|
| 223 |
+
## Model Architecture
|
| 224 |
+
|
| 225 |
+
FastConformer [1] is an optimized version of the Conformer model with 8x depthwise-separable convolutional downsampling. The model is trained using CTC loss. You may find more information on the details of FastConformer here: [Fast-Conformer Model](https://docs.nvidia.com/deeplearning/nemo/user-guide/docs/en/main/asr/models.html#fast-conformer).
|
| 226 |
+
|
| 227 |
+
## Training
|
| 228 |
+
|
| 229 |
+
The NeMo toolkit [3] was used for training the models for over several hundred epochs. These model are trained with this [example script](https://github.com/NVIDIA/NeMo/blob/main/examples/asr/asr_ctc/speech_to_text_ctc_bpe.py) and this [base config](https://github.com/NVIDIA/NeMo/blob/main/examples/asr/conf/fastconformer/fast-conformer_ctc_bpe.yaml).
|
| 230 |
+
|
| 231 |
+
The tokenizers for these models were built using the text transcripts of the train set with this [script](https://github.com/NVIDIA/NeMo/blob/main/scripts/tokenizers/process_asr_text_tokenizer.py).
|
| 232 |
+
|
| 233 |
+
### Datasets
|
| 234 |
+
|
| 235 |
+
The model was trained on 64K hours of English speech collected and prepared by NVIDIA NeMo and Suno teams.
|
| 236 |
+
|
| 237 |
+
The training dataset consists of private subset with 40K hours of English speech plus 24K hours from the following public datasets:
|
| 238 |
+
|
| 239 |
+
- Librispeech 960 hours of English speech
|
| 240 |
+
- Fisher Corpus
|
| 241 |
+
- Switchboard-1 Dataset
|
| 242 |
+
- WSJ-0 and WSJ-1
|
| 243 |
+
- National Speech Corpus (Part 1, Part 6)
|
| 244 |
+
- VCTK
|
| 245 |
+
- VoxPopuli (EN)
|
| 246 |
+
- Europarl-ASR (EN)
|
| 247 |
+
- Multilingual Librispeech (MLS EN) - 2,000 hour subset
|
| 248 |
+
- Mozilla Common Voice (v7.0)
|
| 249 |
+
- People's Speech - 12,000 hour subset
|
| 250 |
+
|
| 251 |
+
## Performance
|
| 252 |
+
|
| 253 |
+
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.
|
| 254 |
+
|
| 255 |
+
The following tables summarizes the performance of the available models in this collection with the CTC decoder. Performances of the ASR models are reported in terms of Word Error Rate (WER%) with greedy decoding.
|
| 256 |
+
|
| 257 |
+
|**Version**|**Tokenizer**|**Vocabulary Size**|**AMI**|**Earnings-22**|**Giga Speech**|**LS test-clean**|**SPGI Speech**|**TEDLIUM-v3**|**Vox Populi**|**Common Voice**|
|
| 258 |
+
|---------|-----------------------|-----------------|---------------|---------------|------------|-----------|-----|-------|------|------|
|
| 259 |
+
| 1.22.0 | SentencePiece Unigram | 1024 | 16.30 | 14.14 | 10.35 | 1.87 | 3.76 | 4.11 | 3.78 | 7.00 | 10.57 |
|
| 260 |
+
|
| 261 |
+
These are greedy WER numbers without external LM. More details on evaluation can be found at [HuggingFace ASR Leaderboard](https://huggingface.co/spaces/hf-audio/open_asr_leaderboard)
|
| 262 |
+
|
| 263 |
+
## NVIDIA Riva: Deployment
|
| 264 |
+
|
| 265 |
+
[NVIDIA Riva](https://developer.nvidia.com/riva), is an accelerated speech AI SDK deployable on-prem, in all clouds, multi-cloud, hybrid, on edge, and embedded.
|
| 266 |
+
Additionally, Riva provides:
|
| 267 |
+
|
| 268 |
+
* 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
|
| 269 |
+
* 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
|
| 270 |
+
* Streaming speech recognition, Kubernetes compatible scaling, and enterprise-grade support
|
| 271 |
+
|
| 272 |
+
Although this model isn’t supported yet by Riva, the [list of supported models is here](https://huggingface.co/models?other=Riva).
|
| 273 |
+
Check out [Riva live demo](https://developer.nvidia.com/riva#demos).
|
| 274 |
+
|
| 275 |
+
## References
|
| 276 |
+
[1] [Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition](https://arxiv.org/abs/2305.05084)
|
| 277 |
+
|
| 278 |
+
[2] [Google Sentencepiece Tokenizer](https://github.com/google/sentencepiece)
|
| 279 |
+
|
| 280 |
+
[3] [NVIDIA NeMo Toolkit](https://github.com/NVIDIA/NeMo)
|
| 281 |
+
|
| 282 |
+
[4] [Suno.ai](https://suno.ai/)
|
| 283 |
+
|
| 284 |
+
[5] [HuggingFace ASR Leaderboard](https://huggingface.co/spaces/hf-audio/open_asr_leaderboard)
|
| 285 |
+
|
| 286 |
+
|
| 287 |
+
## Licence
|
| 288 |
+
|
| 289 |
+
License to use this model is covered by the [CC-BY-4.0](https://creativecommons.org/licenses/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](https://creativecommons.org/licenses/by/4.0/) license.
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