Automatic Speech Recognition
NeMo
Safetensors
Transformers
PyTorch
nemotron3_5_asr
feature-extraction
speech-recognition
cache-aware ASR
streaming-asr
multilingual
speech
audio
FastConformer
RNNT
Parakeet
ASR
NeMo
Eval Results (legacy)
Eval Results
Instructions to use nvidia/nemotron-3.5-asr-streaming-0.6b with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- NeMo
How to use nvidia/nemotron-3.5-asr-streaming-0.6b with NeMo:
import nemo.collections.asr as nemo_asr asr_model = nemo_asr.models.ASRModel.from_pretrained("nvidia/nemotron-3.5-asr-streaming-0.6b") transcriptions = asr_model.transcribe(["file.wav"]) - Transformers
How to use nvidia/nemotron-3.5-asr-streaming-0.6b with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="nvidia/nemotron-3.5-asr-streaming-0.6b")# Load model directly from transformers import AutoProcessor, AutoModel processor = AutoProcessor.from_pretrained("nvidia/nemotron-3.5-asr-streaming-0.6b") model = AutoModel.from_pretrained("nvidia/nemotron-3.5-asr-streaming-0.6b", device_map="auto") - Inference
- Notebooks
- Google Colab
- Kaggle
Add NeMo-Speech.cpp GGUF
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README.md
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##
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```bash
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```
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### Loading the Model
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```python
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import nemo.collections.asr as nemo_asr
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asr_model = nemo_asr.models.ASRModel.from_pretrained(model_name="nvidia/nemotron-3.5-asr-streaming-0.6b")
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```
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### Streaming Inference
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You can use the cache-aware streaming inference script from NeMo - [NeMo/examples/asr/asr_cache_aware_streaming/speech_to_text_cache_aware_streaming_infer.py](https://github.com/NVIDIA-NeMo/NeMo/blob/main/examples/asr/asr_cache_aware_streaming/speech_to_text_cache_aware_streaming_infer.py)
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- `strip_lang_tags=false` (keep): the tag is left in the output, so you can read the detected language directly from each utterance — useful for mixed-language traffic and language labeling.
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- `strip_lang_tags=true` (remove): the tag is stripped, leaving only the clean transcript text — useful when you only need the spoken words.
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### Setting up Streaming Configuration
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Latency is defined by the `att_context_size` param, where att_context_size = `{num_frames_left_context, num_frame_right_context}`, all measured in **80ms frames**:
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---
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## How to Use this Model
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There are several ways to use this model. Choose the one that fits your needs.
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### Run locally with NeMo-Speech.cpp
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[NeMo-Speech.cpp](https://github.com/NVIDIA/NeMo-Speech.cpp) provides a
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lightweight native C++ runtime for local inference with
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this model. After [installing the runtime](https://github.com/NVIDIA/NeMo-Speech.cpp#installation):
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```bash
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hf download nvidia/nemotron-3.5-asr-streaming-0.6b \
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nemotron-3.5-asr-streaming-0.6b.q8_0.gguf \
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--local-dir models
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nemo-speech transcribe audio.wav \
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--model models/nemotron-3.5-asr-streaming-0.6b.q8_0.gguf \
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--language en-US
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```
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Use another supported locale or `--language auto` for automatic language
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detection. See the [NeMo-Speech.cpp documentation](https://github.com/NVIDIA/NeMo-Speech.cpp)
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for more details.
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### NVIDIA NeMo
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To train, fine-tune or perform inference with this model, install [NVIDIA NeMo](https://github.com/NVIDIA/NeMo) [\[4\]](#ref-4) after installing Python 3.11 or later, Cython, and a recent PyTorch version.
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```bash
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apt-get update && apt-get install -y libsndfile1 ffmpeg
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pip install Cython packaging
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pip install git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]
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```
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#### Loading the Model
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```python
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import nemo.collections.asr as nemo_asr
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asr_model = nemo_asr.models.ASRModel.from_pretrained(model_name="nvidia/nemotron-3.5-asr-streaming-0.6b")
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```
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#### Streaming Inference
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You can use the cache-aware streaming inference script from NeMo - [NeMo/examples/asr/asr_cache_aware_streaming/speech_to_text_cache_aware_streaming_infer.py](https://github.com/NVIDIA-NeMo/NeMo/blob/main/examples/asr/asr_cache_aware_streaming/speech_to_text_cache_aware_streaming_infer.py)
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- `strip_lang_tags=false` (keep): the tag is left in the output, so you can read the detected language directly from each utterance — useful for mixed-language traffic and language labeling.
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- `strip_lang_tags=true` (remove): the tag is stripped, leaving only the clean transcript text — useful when you only need the spoken words.
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#### Setting up Streaming Configuration
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Latency is defined by the `att_context_size` param, where att_context_size = `{num_frames_left_context, num_frame_right_context}`, all measured in **80ms frames**:
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version https://git-lfs.github.com/spec/v1
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oid sha256:a5c435f294eea8f88ce68dd27b8c3bfea7f777cb2fbba04fcd30eaa555f429ae
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size 741548352
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