Commit
·
ef9aeeb
1
Parent(s):
2f0d418
correct readme
Browse files
README.md
CHANGED
|
@@ -17,3 +17,53 @@ Learning, Semi-Supervised Learning and Interpretation](https://arxiv.org/abs/210
|
|
| 17 |
**Authors**: *Changhan Wang, Morgane Riviere, Ann Lee, Anne Wu, Chaitanya Talnikar, Daniel Haziza, Mary Williamson, Juan Pino, Emmanuel Dupoux* from *Facebook AI*
|
| 18 |
|
| 19 |
See the official website for more information, [here](https://github.com/facebookresearch/voxpopuli/)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 17 |
**Authors**: *Changhan Wang, Morgane Riviere, Ann Lee, Anne Wu, Chaitanya Talnikar, Daniel Haziza, Mary Williamson, Juan Pino, Emmanuel Dupoux* from *Facebook AI*
|
| 18 |
|
| 19 |
See the official website for more information, [here](https://github.com/facebookresearch/voxpopuli/)
|
| 20 |
+
|
| 21 |
+
|
| 22 |
+
# Usage for inference
|
| 23 |
+
|
| 24 |
+
In the following it is shown how the model can be used in inference on a sample of the [Common Voice dataset](https://commonvoice.mozilla.org/en/datasets)
|
| 25 |
+
|
| 26 |
+
```python
|
| 27 |
+
#!/usr/bin/env python3
|
| 28 |
+
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
|
| 29 |
+
from datasets import load_dataset
|
| 30 |
+
import torchaudio
|
| 31 |
+
import torch
|
| 32 |
+
|
| 33 |
+
# resample audio
|
| 34 |
+
|
| 35 |
+
# load model & processor
|
| 36 |
+
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-10k-voxpopuli-ft-cs")
|
| 37 |
+
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-10k-voxpopuli-ft-cs")
|
| 38 |
+
|
| 39 |
+
# load dataset
|
| 40 |
+
ds = load_dataset("common_voice", "cs", split="validation[:1%]")
|
| 41 |
+
|
| 42 |
+
# common voice does not match target sampling rate
|
| 43 |
+
common_voice_sample_rate = 48000
|
| 44 |
+
target_sample_rate = 16000
|
| 45 |
+
|
| 46 |
+
resampler = torchaudio.transforms.Resample(common_voice_sample_rate, target_sample_rate)
|
| 47 |
+
|
| 48 |
+
|
| 49 |
+
# define mapping fn to read in sound file and resample
|
| 50 |
+
def map_to_array(batch):
|
| 51 |
+
speech, _ = torchaudio.load(batch["path"])
|
| 52 |
+
speech = resampler(speech)
|
| 53 |
+
batch["speech"] = speech[0]
|
| 54 |
+
return batch
|
| 55 |
+
|
| 56 |
+
|
| 57 |
+
# load all audio files
|
| 58 |
+
ds = ds.map(map_to_array)
|
| 59 |
+
|
| 60 |
+
# run inference on the first 5 data samples
|
| 61 |
+
inputs = processor(ds[:5]["speech"], sampling_rate=target_sample_rate, return_tensors="pt", padding=True)
|
| 62 |
+
|
| 63 |
+
# inference
|
| 64 |
+
logits = model(**inputs).logits
|
| 65 |
+
predicted_ids = torch.argmax(logits, axis=-1)
|
| 66 |
+
|
| 67 |
+
print(processor.batch_decode(predicted_ids))
|
| 68 |
+
```
|
| 69 |
+
|