README.md

---
language:
- km
license: apache-2.0
tags:
- automatic-speech-recognition
- openslr
- robust-speech-event
- km
- generated_from_trainer
- hf-asr-leaderboard
datasets:
- openslr
model-index:
- name: wav2vec2-xls-r-1b-km
  results:
  - task:
      name: Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: OpenSLR km
      type: openslr
      args: km
    metrics:
    - name: Test WER
      type: wer
      value: 32.13
    - name: Test CER
      type: cer
      value: 9.35
  - task:
      name: Automatic Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: Robust Speech Event - Dev Data
      type: speech-recognition-community-v2/dev_data
      args: km
    metrics:
    - name: Test WER
      type: wer
      value: 32.13
    - name: Test CER
      type: cer
      value: 9.35
---
# 

This model is a fine-tuned version of [facebook/wav2vec2-xls-r-1b](https://huggingface.co/facebook/wav2vec2-xls-r-1b) on the openslr dataset.
It achieves the following results on the evaluation set:
- Loss: 0.4239
- Wer: 0.4221

# Evaluation results on OpenSLR "test" (self-split 10%) (Running ./eval.py):
- WER: 0.4490281634272114
- CER: 0.12198285179047481

# Evaluation results on OpenSLR "test" with LM ngram (self-split 10%) (Running ./eval.py):
- WER: 0.32130107100357
- CER: 0.09345053678218891

# Note
- Since this dataset is small (4 hours of voice recording), we decided not to train that for too long to avoid overfitting and under-generalization.
- This model performs worse than its 300M-variant. Probably, we don't explore the hyper-parameter enough?


## Installation
Install the following libraries on top of HuggingFace Transformers for the supports of language model.
```
pip install pyctcdecode
pip install https://github.com/kpu/kenlm/archive/master.zip
```

## Usage 

**Approach 1:** Using HuggingFace's pipeline, this will cover everything end-to-end from raw audio input to text output.
```python
from transformers import pipeline

# Load the model
pipe = pipeline(model="vitouphy/wav2vec2-xls-r-300m-khmer")

# Process raw audio
output = pipe("sound_file.wav", chunk_length_s=10, stride_length_s=(4, 2))
```

**Approach 2:** More custom way to predict phonemes.
```python
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC 
import librosa
import torch

# load model and processor
processor = Wav2Vec2Processor.from_pretrained("vitouphy/wav2vec2-xls-r-300m-khmer")
model = Wav2Vec2ForCTC.from_pretrained("vitouphy/wav2vec2-xls-r-300m-khmer")

# Read and process the input
speech_array, sampling_rate = librosa.load("sound_file.wav", sr=16_000)
inputs = processor(speech_array, sampling_rate=16_000, return_tensors="pt", padding=True)

with torch.no_grad():
    logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits

predicted_ids = torch.argmax(logits, axis=-1)      
predicted_sentences = processor.batch_decode(predicted_ids)
print(predicted_sentences)
```

## Intended uses & limitations

The data used for this model is only around 4 hours of recordings. 
- We split into 80/10/10. Hence, the training hour is 3.2 hours, which is very very small.
- Yet, its performance is not too bad. Quite interesting for such small dataset, actually. You can try it out.
- Its limitation is:
  - Rare characters, e.g. ឬស្សី ឪឡឹក 
  - Speech needs to be clear and articulate.
- More data to cover more vocabulary and character may help improve this system.


## Training procedure

### Training hyperparameters

The following hyperparameters were used during training:
- learning_rate: 1e-05
- train_batch_size: 8
- eval_batch_size: 8
- seed: 42
- gradient_accumulation_steps: 4
- total_train_batch_size: 32
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 2000
- num_epochs: 75
- mixed_precision_training: Native AMP

### Training results

| Training Loss | Epoch | Step | Validation Loss | Wer    |
|:-------------:|:-----:|:----:|:---------------:|:------:|
| 3.5671        | 5.47  | 400  | 12.0218         | 1.0    |
| 3.5159        | 10.95 | 800  | 10.6337         | 1.0    |
| 2.4543        | 16.43 | 1200 | 1.8256          | 0.9839 |
| 1.9437        | 21.91 | 1600 | 1.1237          | 0.9173 |
| 1.696         | 27.39 | 2000 | 0.8246          | 0.7700 |
| 1.5342        | 32.87 | 2400 | 0.6433          | 0.6594 |
| 1.4509        | 38.35 | 2800 | 0.5500          | 0.5787 |
| 1.3478        | 43.83 | 3200 | 0.5070          | 0.4907 |
| 1.3096        | 49.31 | 3600 | 0.4692          | 0.4726 |
| 1.2532        | 54.79 | 4000 | 0.4448          | 0.4479 |
| 1.2291        | 60.27 | 4400 | 0.4374          | 0.4366 |
| 1.196         | 65.75 | 4800 | 0.4314          | 0.4310 |
| 1.1862        | 71.23 | 5200 | 0.4239          | 0.4221 |


### Framework versions

- Transformers 4.17.0.dev0
- Pytorch 1.10.2+cu102
- Datasets 1.18.2.dev0
- Tokenizers 0.11.0