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: current WER table, eval results (model-index), tags; list AutoModelForSpeechSeq2Seq first in auto_map
Browse files- README.md +172 -18
- config.json +1 -1
README.md
CHANGED
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@@ -18,11 +18,164 @@ pipeline_tag: automatic-speech-recognition
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tags:
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- asr
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- speech-recognition
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- audio
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- qwen
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- granite-speech
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- lora
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library_name: transformers
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---
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# Tiny Audio
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@@ -30,8 +183,8 @@ library_name: transformers
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**A tiny, hackable, open-source speech LLM.**
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It transcribes English with punctuation, capitalization, word timestamps, and speaker labels: 1.8%
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-
WER on LibriSpeech test-clean and
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-
[Tiny Audio](https://github.com/alexkroman/tiny-audio).
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**[Try the live demo](https://huggingface.co/spaces/mazesmazes/tiny-audio)** ·
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**[Train your own](https://github.com/alexkroman/tiny-audio)** ·
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@@ -58,25 +211,26 @@ the model writes punctuation, capitalization, and numbers itself.
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## Benchmarks
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Word error rate (%, lower is better) on
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[Tiny Audio eval harness](https://github.com/alexkroman/tiny-audio/tree/main/scripts/eval)
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(`ta eval`) after text normalization.
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-
| Dataset
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-
| ---------------------- | -------: |
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-
| LibriSpeech test-clean
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-
| SPGISpeech
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-
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-
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-
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-
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-
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-
| AMI (IHM)
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| GigaSpeech
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| Earnings22 †
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| People's Speech
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| AMI (SDM)
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| **Mean (12 sets)**
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† Held out: no data from this source was used in training.
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tags:
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- asr
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- speech-recognition
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- speech-to-text
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- audio
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+
- speech-llm
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- word-timestamps
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- speaker-diarization
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- qwen
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- granite-speech
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- lora
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library_name: transformers
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metrics:
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- wer
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model-index:
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- name: tiny-audio
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results:
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "LibriSpeech test-clean, subset of up to 1,000 samples"
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type: openslr/librispeech_asr
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config: clean
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split: test
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metrics:
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- type: wer
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value: 1.84
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "LibriSpeech test-other, subset of up to 1,000 samples"
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type: openslr/librispeech_asr
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config: other
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split: test
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metrics:
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- type: wer
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value: 6.38
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "SPGISpeech, subset of up to 1,000 samples"
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type: kensho/spgispeech
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config: test
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split: test
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metrics:
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- type: wer
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value: 2.29
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "TED-LIUM, subset of up to 1,000 samples"
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type: sanchit-gandhi/tedlium-data
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config: default
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split: test
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metrics:
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- type: wer
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value: 3.71
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "LoquaciousSet (held out), subset of up to 1,000 samples"
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type: speechbrain/LoquaciousSet
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config: small
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split: test
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metrics:
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- type: wer
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value: 6.2
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "VoxPopuli, subset of up to 1,000 samples"
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type: facebook/voxpopuli
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config: en
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split: test
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metrics:
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- type: wer
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value: 7.11
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "Common Voice 17, subset of up to 1,000 samples"
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type: fixie-ai/common_voice_17_0
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config: en
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split: test
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metrics:
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- type: wer
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value: 7.18
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "AMI (IHM), subset of up to 1,000 samples"
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type: edinburghcstr/ami
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config: ihm
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split: test
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metrics:
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- type: wer
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value: 8.99
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "GigaSpeech (dev), subset of up to 1,000 samples"
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type: fixie-ai/gigaspeech
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config: dev
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split: dev
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metrics:
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- type: wer
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value: 9.06
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "Earnings22 (held out), subset of up to 1,000 samples"
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type: sanchit-gandhi/earnings22_robust_split
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config: default
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split: test
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metrics:
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- type: wer
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value: 10.58
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name: WER (normalized, subset of up to 1,000 samples)
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "People's Speech, subset of up to 1,000 samples"
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type: fixie-ai/peoples_speech
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config: clean
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split: test
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metrics:
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- type: wer
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value: 17.59
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name: WER (normalized, subset of up to 1,000 samples)
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+
- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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dataset:
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name: "AMI (SDM), subset of up to 1,000 samples"
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type: edinburghcstr/ami
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config: sdm
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split: test
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metrics:
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- type: wer
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value: 23.53
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name: WER (normalized, subset of up to 1,000 samples)
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---
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# Tiny Audio
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**A tiny, hackable, open-source speech LLM.**
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It transcribes English with punctuation, capitalization, word timestamps, and speaker labels: 1.8%
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+
WER on LibriSpeech test-clean and 7.4% across 12 benchmarks (11,822 samples pooled). Built and
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trained with [Tiny Audio](https://github.com/alexkroman/tiny-audio).
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**[Try the live demo](https://huggingface.co/spaces/mazesmazes/tiny-audio)** ·
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**[Train your own](https://github.com/alexkroman/tiny-audio)** ·
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## Benchmarks
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Word error rate (%, lower is better) on 11,822 samples (up to 1,000 per dataset), scored with the
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[Tiny Audio eval harness](https://github.com/alexkroman/tiny-audio/tree/main/scripts/eval)
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(`ta eval`) after text normalization.
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+
| Dataset | WER |
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| --------------------------- | -------: |
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+
| LibriSpeech test-clean | 1.84 |
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+
| SPGISpeech | 2.29 |
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+
| TED-LIUM | 3.71 |
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| LoquaciousSet † | 6.20 |
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| LibriSpeech test-other | 6.38 |
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| VoxPopuli | 7.11 |
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| Common Voice | 7.18 |
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| AMI (IHM) | 8.99 |
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| GigaSpeech | 9.06 |
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| Earnings22 † | 10.58 |
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| People's Speech | 17.59 |
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| AMI (SDM) | 23.53 |
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| **Mean (12 sets)** | **8.71** |
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| **Pooled (11,822 samples)** | **7.42** |
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† Held out: no data from this source was used in training.
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config.json
CHANGED
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"audio_token": "<audio>",
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"auto_map": {
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"AutoConfig": "asr_config.ASRConfig",
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-
"AutoModel": "asr_modeling.ASRModel",
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"AutoModelForSpeechSeq2Seq": "asr_modeling.ASRModel",
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"AutoProcessor": "asr_processing.ASRProcessor"
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},
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"custom_pipelines": {
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"audio_token": "<audio>",
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"auto_map": {
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"AutoConfig": "asr_config.ASRConfig",
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"AutoModelForSpeechSeq2Seq": "asr_modeling.ASRModel",
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"AutoModel": "asr_modeling.ASRModel",
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"AutoProcessor": "asr_processing.ASRProcessor"
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},
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"custom_pipelines": {
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