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---
license: apache-2.0
base_model: openai/whisper-large-v3
tags:
- generated_from_trainer
metrics:
- wer
model-index:
- name: Hibiki_ASR_Phonemizer
results: []
language:
- ja
---
# Hibiki ASR Phonemizer
This model is a Phoneme Level Speech Recognition network, originally a fine-tuned version of [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) on a
mixture of Different Japanese datasets.
it can detect, transcribe and do the following:
- non-speech sounds such as gasp, erotic moans, laughter, etc.
- adding punctuations more faithfully.
a Grapheme decoder head (i.e outputting normal Japanese) will probably be trained as well. Though going directly from audio to Phonemes will result in a
more accurate representation for Japanese.
evaluation set:
- Loss: 0.2186
- Wer: 21.6707
## Inference and Post-proc
```python
# this function was borrowed and modified from Aaron Yinghao Li, the Author of StyleTTS paper.
from datasets import Dataset, Audio
from transformers import WhisperProcessor, WhisperForConditionalGeneration
import jaconv
kana_mapper = dict([
("ゔぁ","ba"),
.
.
.
etc. # Take a look at the Notebook for the whole code
("ぉ"," o"),
("ゎ"," ɯa"),
("ぉ"," o"),
("を","o")
])
def post_fix(text):
orig = text
for k, v in kana_mapper.items():
text = text.replace(k, v)
return text
processor = WhisperProcessor.from_pretrained("openai/whisper-large-v3")
model = WhisperForConditionalGeneration.from_pretrained("Respair/Hibiki_ASR_Phonemizer").to("cuda:0")
forced_decoder_ids = processor.get_decoder_prompt_ids(task="transcribe", language='japanese')
import re
sample = Dataset.from_dict({"audio": ["/content/kl_chunk1987.wav"]}).cast_column("audio", Audio(16000))
sample = sample[0]['audio']
# Ensure the input features are on the same device as the model
input_features = processor(sample["array"], sampling_rate=sample["sampling_rate"], return_tensors="pt").input_features.to("cuda:0")
# generate token ids
predicted_ids = model.generate(input_features,forced_decoder_ids=forced_decoder_ids, repetition_penalty=1.2)
# decode token ids to text
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
# You can add your final adjustments here, it's better to write a dict though, but I'm just giving you a quick demonstration here.
if ' neɽitai ' in transcription[0]:
transcription[0] = transcription[0].replace(' neɽitai ', "naɽitai")
if 'harɯdʑisama' in transcription[0]:
transcription[0] = transcription[0].replace('harɯdʑisama', "arɯdʑisama")
if "ki ni ɕinai" in transcription[0]:
transcription[0] = re.sub(r'(?<!\s)ki ni ɕinai', r' ki ni ɕinai', transcription[0])
if 'ʔt' in transcription[0]:
transcription[0] = re.sub(r'(?<!\s)ʔt', r'ʔt', transcription[0])
if 'de aɽoɯ' in transcription[0]:
transcription[0] = re.sub(r'(?<!\s)de aɽoɯ', r' de aɽoɯ', transcription[0])
post_fix(jaconv.kata2hira(transcription[0].lstrip())) # Ensuring the model won't hallucinate and return kana
```
the Full code -> [Notebook](https://colab.research.google.com/drive/13tx8WKzkvePFdtKU4WUE_iYyYCqTY8dZ#scrollTo=5XqUs-sPdT79)
## Intended uses & limitations
No restrictions is imposed by me, but proceed at your own risk, The User (You) are entirely responisble for their actions.
## Training and evaluation data
- Japanese Common Voice 17
- ehehe Corpus
- Custom Game and Anime dataset (around 8 hours)
## Training procedure
### Training hyperparameters
The following hyperparameters were used during training:
- learning_rate: 1e-05
- train_batch_size: 24
- eval_batch_size: 8
- seed: 42
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 500
- training_steps: 6000
### Training results
| Training Loss | Epoch | Step | Validation Loss | Wer |
|:-------------:|:------:|:----:|:---------------:|:-------:|
| 0.2101 | 0.8058 | 1000 | 0.2090 | 30.1840 |
| 0.1369 | 1.6116 | 2000 | 0.1837 | 27.6756 |
| 0.0838 | 2.4174 | 3000 | 0.1829 | 26.4036 |
| 0.0454 | 3.2232 | 4000 | 0.1922 | 20.9549 |
| 0.0434 | 4.0290 | 5000 | 0.2072 | 20.8898 |
| 0.021 | 4.8348 | 6000 | 0.2186 | 21.6707 |
### Compute and Duration
- 1x A100(40G)
- 64gb RAM
- BF16
- 14hrs
### Framework versions
- Transformers 4.41.1
- Pytorch 2.4.0+cu121
- Datasets 2.19.1
- Tokenizers 0.19.1 |