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---
license: apache-2.0
datasets:
- common_voice
language:
- ja
tags:
- audio
---
# Fine-tuned Japanese Whisper model for speech recognition using whisper-small
Fine-tuned [openai/whisper-small](https://huggingface.co/openai/whisper-small) on Japanese using [Common Voice](https://commonvoice.mozilla.org/ja/datasets), [JVS](https://sites.google.com/site/shinnosuketakamichi/research-topics/jvs_corpus) and [JSUT](https://sites.google.com/site/shinnosuketakamichi/publication/jsut).
When using this model, make sure that your speech input is sampled at 16kHz.
## Usage
The model can be used directly as follows.
```python
from transformers import WhisperForConditionalGeneration, WhisperProcessor
from datasets import load_dataset
import librosa
import torch
LANG_ID = "ja"
MODEL_ID = "Ivydata/whisper-small-japanese"
SAMPLES = 10
test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
processor = WhisperProcessor.from_pretrained(MODEL_ID)
model = WhisperForConditionalGeneration.from_pretrained(MODEL_ID)
model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(
language="ja", task="transcribe"
)
model.config.suppress_tokens = []
# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
batch["speech"] = speech_array
batch["sentence"] = batch["sentence"].upper()
batch["sampling_rate"] = sampling_rate
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
sample = test_dataset[0]
input_features = processor(sample["speech"], sampling_rate=sample["sampling_rate"], return_tensors="pt").input_features
predicted_ids = model.generate(input_features)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=False)
# ['<|startoftranscript|><|ja|><|transcribe|><|notimestamps|>ζ¨ζγγγ«ι»θ©±γθ²Έγγ¦γγγγΎγγγ<|endoftext|>']
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
# ['ζ¨ζγγγ«ι»θ©±γθ²Έγγ¦γγγγΎγγγ']
``` |