Automatic Speech Recognition
Transformers
Safetensors
English
asr_model
asr
speech-recognition
speech-to-text
audio
speech-llm
word-timestamps
speaker-diarization
qwen
granite-speech
lora
custom_code
Eval Results (legacy)
Instructions to use mazesmazes/tiny-audio with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use mazesmazes/tiny-audio with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="mazesmazes/tiny-audio", trust_remote_code=True)# pip install -U transformers accelerate # Load model directly from transformers import AutoModelForSpeechSeq2Seq model = AutoModelForSpeechSeq2Seq.from_pretrained("mazesmazes/tiny-audio", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
Model card: long-audio and diarization limits, credit Nemotron and Qwen aligner
Browse files
README.md
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@@ -207,7 +207,8 @@ The training recipe is [`configs/experiments/granite_qwen_frozen.yaml`](https://
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- **English only**: Not trained on other languages.
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- **Sample rate**: Expects 16kHz audio (other rates are resampled automatically).
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- **Audio length**:
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- **Accuracy**: May degrade on:
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- Far-field and overlapping speech (see AMI SDM)
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- Noisy or low-quality audio
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- [Granite Speech](https://huggingface.co/ibm-granite/granite-speech-5.0-470m-turboctc) for the audio encoder
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- [Qwen3.5](https://huggingface.co/Qwen/Qwen3.5-2B) for the language model
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- The LibriHeavy, People's Speech, Common Voice, GigaSpeech, SPGISpeech, VoxPopuli, AMI, and TED-LIUM teams for training data
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## License
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- **English only**: Not trained on other languages.
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- **Sample rate**: Expects 16kHz audio (other rates are resampled automatically).
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- **Audio length**: A plain `pipe(audio)` call decodes the clip in one pass and works best up to about 19 seconds (the training length). For longer audio pass `return_timestamps=True` or `return_speakers=True`, which transcribe in 8-18 s chunks automatically.
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- **Speaker diarization**: At most 8 speakers per recording. The count is detected automatically; `num_speakers` and `max_speakers` can cap it, but `min_speakers` is not supported (passing it raises a `ValueError`). Needs transformers `main` until a release includes Nemotron-3-Diarization.
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- **Accuracy**: May degrade on:
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- Far-field and overlapping speech (see AMI SDM)
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- Noisy or low-quality audio
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- [Granite Speech](https://huggingface.co/ibm-granite/granite-speech-5.0-470m-turboctc) for the audio encoder
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- [Qwen3.5](https://huggingface.co/Qwen/Qwen3.5-2B) for the language model
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- [Qwen3-ForcedAligner](https://huggingface.co/Qwen/Qwen3-ForcedAligner-0.6B-hf) for word timestamps
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- [Nemotron-3-Diarization](https://huggingface.co/nvidia/Nemotron-3-Diarization) for speaker diarization
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- The LibriHeavy, People's Speech, Common Voice, GigaSpeech, SPGISpeech, VoxPopuli, AMI, and TED-LIUM teams for training data
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## License
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