openslr/librispeech_asr
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How to use speech-seq2seq/wav2vec2-2-gpt2-no-adapter with Transformers:
# Use a pipeline as a high-level helper
from transformers import pipeline
pipe = pipeline("automatic-speech-recognition", model="speech-seq2seq/wav2vec2-2-gpt2-no-adapter") # Load model directly
from transformers import AutoTokenizer, AutoModelForSpeechSeq2Seq
tokenizer = AutoTokenizer.from_pretrained("speech-seq2seq/wav2vec2-2-gpt2-no-adapter")
model = AutoModelForSpeechSeq2Seq.from_pretrained("speech-seq2seq/wav2vec2-2-gpt2-no-adapter")YAML Metadata Error:"model-index[0].name" is not allowed to be empty
This model was trained from scratch on the librispeech_asr dataset. It achieves the following results on the evaluation set:
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The following hyperparameters were used during training:
| Training Loss | Epoch | Step | Validation Loss | Wer |
|---|---|---|---|---|
| 4.7015 | 0.28 | 500 | 5.3313 | 1.9454 |
| 4.7239 | 0.56 | 1000 | 5.1316 | 1.9288 |
| 4.6686 | 0.84 | 1500 | 4.8812 | 1.9646 |
| 4.0138 | 1.12 | 2000 | 4.8274 | 1.8905 |
| 3.6314 | 1.4 | 2500 | 3.8913 | 1.7298 |
| 1.9511 | 1.68 | 3000 | 2.3486 | 1.3674 |
| 1.212 | 1.96 | 3500 | 1.6223 | 1.1877 |
| 0.8092 | 2.24 | 4000 | 1.3949 | 1.1049 |
| 0.497 | 2.52 | 4500 | 1.2544 | 1.0749 |
| 0.4401 | 2.8 | 5000 | 1.1277 | 1.0334 |