parakeet-tdt-0.6b-v3
nvidia/parakeet-tdt-0.6b-v3
- 作者
- nvidia
- 任务
- 语音识别
- 下载量
- 574,141
- 点赞数
- 1,255
- 趋势分
- 25
- 许可证
- cc-by-4.0
- 收录时间
- 2026/11/4
license: cc-by-4.0 track_downloads: true language:
- en
- es
- fr
- de
- bg
- hr
- cs
- da
- nl
- et
- fi
- el
- hu
- it
- lv
- lt
- mt
- pl
- pt
- ro
- sk
- sl
- sv
- ru
- uk pipeline_tag: automatic-speech-recognition library_name: transformers datasets:
- nvidia/Granary
- nemo/asr-set-3.0 tags:
- automatic-speech-recognition
- speech
- audio
- Transducer
- Transformer
- TDT
- FastConformer
- Conformer
- pytorch
- NeMo
- hf-asr-leaderboard
- Transformers widget:
- example_title: Librispeech sample 1 src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
- example_title: Librispeech sample 2 src: https://cdn-media.huggingface.co/speech_samples/sample2.flac metrics:
- wer model-index:
- name: parakeet-tdt-0.6b-v3
results:
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: AMI (Meetings test)
type: edinburghcstr/ami
config: ihm
split: test
args:
language: en
metrics:
- type: wer value: 11.31 name: Test WER
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: Earnings-22
type: revdotcom/earnings22
split: test
args:
language: en
metrics:
- type: wer value: 11.42 name: Test WER
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: GigaSpeech
type: speechcolab/gigaspeech
split: test
args:
language: en
metrics:
- type: wer value: 9.59 name: Test WER
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: LibriSpeech (clean)
type: librispeech_asr
config: other
split: test
args:
language: en
metrics:
- type: wer value: 1.93 name: Test WER
- type: wer value: 3.59 name: Test WER
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: SPGI Speech
type: kensho/spgispeech
config: test
split: test
args:
language: en
metrics:
- type: wer value: 3.97 name: Test WER
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: tedlium-v3
type: LIUM/tedlium
config: release1
split: test
args:
language: en
metrics:
- type: wer value: 2.75 name: Test WER
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: Vox Populi
type: facebook/voxpopuli
config: en
split: test
args:
language: en
metrics:
- type: wer value: 6.14 name: Test WER
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: bg_bg
split: test
args:
language: bg
metrics:
- type: wer value: 12.64 name: Test WER (Bg)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: cs_cz
split: test
args:
language: cs
metrics:
- type: wer value: 11.01 name: Test WER (Cs)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: da_dk
split: test
args:
language: da
metrics:
- type: wer value: 18.41 name: Test WER (Da)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: de_de
split: test
args:
language: de
metrics:
- type: wer value: 5.04 name: Test WER (De)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: el_gr
split: test
args:
language: el
metrics:
- type: wer value: 20.7 name: Test WER (El)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: en_us
split: test
args:
language: en
metrics:
- type: wer value: 4.85 name: Test WER (En)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: es_419
split: test
args:
language: es
metrics:
- type: wer value: 3.45 name: Test WER (Es)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: et_ee
split: test
args:
language: et
metrics:
- type: wer value: 17.73 name: Test WER (Et)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: fi_fi
split: test
args:
language: fi
metrics:
- type: wer value: 13.21 name: Test WER (Fi)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: fr_fr
split: test
args:
language: fr
metrics:
- type: wer value: 5.15 name: Test WER (Fr)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: hr_hr
split: test
args:
language: hr
metrics:
- type: wer value: 12.46 name: Test WER (Hr)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: hu_hu
split: test
args:
language: hu
metrics:
- type: wer value: 15.72 name: Test WER (Hu)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: it_it
split: test
args:
language: it
metrics:
- type: wer value: 3 name: Test WER (It)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: lt_lt
split: test
args:
language: lt
metrics:
- type: wer value: 20.35 name: Test WER (Lt)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: lv_lv
split: test
args:
language: lv
metrics:
- type: wer value: 22.84 name: Test WER (Lv)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: mt_mt
split: test
args:
language: mt
metrics:
- type: wer value: 20.46 name: Test WER (Mt)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: nl_nl
split: test
args:
language: nl
metrics:
- type: wer value: 7.48 name: Test WER (Nl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: pl_pl
split: test
args:
language: pl
metrics:
- type: wer value: 7.31 name: Test WER (Pl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: pt_br
split: test
args:
language: pt
metrics:
- type: wer value: 4.76 name: Test WER (Pt)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: ro_ro
split: test
args:
language: ro
metrics:
- type: wer value: 12.44 name: Test WER (Ro)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: ru_ru
split: test
args:
language: ru
metrics:
- type: wer value: 5.51 name: Test WER (Ru)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: sk_sk
split: test
args:
language: sk
metrics:
- type: wer value: 8.82 name: Test WER (Sk)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: sl_si
split: test
args:
language: sl
metrics:
- type: wer value: 24.03 name: Test WER (Sl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: sv_se
split: test
args:
language: sv
metrics:
- type: wer value: 15.08 name: Test WER (Sv)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: FLEURS
type: google/fleurs
config: uk_ua
split: test
args:
language: uk
metrics:
- type: wer value: 6.79 name: Test WER (Uk)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: spanish
split: test
args:
language: es
metrics:
- type: wer value: 4.39 name: Test WER (Es)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: french
split: test
args:
language: fr
metrics:
- type: wer value: 4.97 name: Test WER (Fr)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: italian
split: test
args:
language: it
metrics:
- type: wer value: 10.08 name: Test WER (It)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: dutch
split: test
args:
language: nl
metrics:
- type: wer value: 12.78 name: Test WER (Nl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: polish
split: test
args:
language: pl
metrics:
- type: wer value: 7.28 name: Test WER (Pl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: Multilingual LibriSpeech
type: facebook/multilingual_librispeech
config: portuguese
split: test
args:
language: pt
metrics:
- type: wer value: 7.5 name: Test WER (Pt)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: de
split: test
args:
language: de
metrics:
- type: wer value: 4.84 name: Test WER (De)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: en
split: test
args:
language: en
metrics:
- type: wer value: 6.8 name: Test WER (En)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: es
split: test
args:
language: es
metrics:
- type: wer value: 3.41 name: Test WER (Es)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: et
split: test
args:
language: et
metrics:
- type: wer value: 22.04 name: Test WER (Et)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: fr
split: test
args:
language: fr
metrics:
- type: wer value: 6.05 name: Test WER (Fr)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: it
split: test
args:
language: it
metrics:
- type: wer value: 3.69 name: Test WER (It)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: lv
split: test
args:
language: lv
metrics:
- type: wer value: 38.36 name: Test WER (Lv)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: nl
split: test
args:
language: nl
metrics:
- type: wer value: 6.5 name: Test WER (Nl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: pt
split: test
args:
language: pt
metrics:
- type: wer value: 3.96 name: Test WER (Pt)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: ru
split: test
args:
language: ru
metrics:
- type: wer value: 3 name: Test WER (Ru)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: sl
split: test
args:
language: sl
metrics:
- type: wer value: 31.8 name: Test WER (Sl)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: sv
split: test
args:
language: sv
metrics:
- type: wer value: 20.16 name: Test WER (Sv)
- task:
type: Automatic Speech Recognition
name: automatic-speech-recognition
dataset:
name: CoVoST2
type: covost2
config: uk
split: test
args:
language: uk
metrics:
- type: wer value: 5.1 name: Test WER (Uk)
- task:
type: automatic-speech-recognition
name: Automatic Speech Recognition
dataset:
name: AMI (Meetings test)
type: edinburghcstr/ami
config: ihm
split: test
args:
language: en
metrics:
🦜 parakeet-tdt-0.6b-v3: Multilingual Speech-to-Text Model
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
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
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)
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))
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)
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):
| 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.
Huggingface Open-ASR-Leaderboard
| 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]
Noise Robustness
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% |
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
[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. |
内容来源：Hugging Face 模型卡
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