---
library_name: transformers
pipeline_tag: automatic-speech-recognition
tags:
  - speech
  - asr
  - ctc
  - wav2vec2
  - common-voice
  - onnx
  - sagemaker
  - huggingface
  - transformers
  - jiwer
datasets:
  - mozilla-foundation/common_voice_17_0
base_model:
  - facebook/wav2vec2-base-960h
license: other
language: en
metrics:
  - wer
  - cer
---

# Model Card for **ASR** (CTC-based ASR on English)

<!-- Provide a quick summary of what the model is/does. -->
This repository contains an end‑to‑end **Automatic Speech Recognition (ASR)** pipeline built around Hugging Face Transformers. The default configuration fine‑tunes **`facebook/wav2vec2-base-960h`** with a **CTC** head on  50k subsample of **Common Voice 17.0 (English)** and provides scripts to **train, evaluate, export to ONNX, and deploy on AWS SageMaker**. It also includes a robust audio loading stack (FFmpeg preferred, with fallbacks) and utilities for text normalization and evaluation (WER/CER).

## Model Details

### Model Description

- **Developed by:** Amirhossein Yousefi (GitHub: `@amirhossein-yousefi`)
- **Funded by :** Not specified
- **Shared by :** Amirhossein Yousefi
- **Model type:** CTC-based ASR using Transformers (**Wav2Vec2ForCTC**)
- **Language(s) (NLP):** English (`en`)
- **License:** Base model is Apache-2.0; repository/fine-tuned weights license not explicitly stated here (treat as **other** until clarified)
- **Finetuned from model :** `facebook/wav2vec2-base-960h`

> The training/evaluation pipeline uses Hugging Face `transformers`, `datasets`, and `jiwer` and includes scripts for inference and SageMaker deployment.

### Model Sources 

- **Repository:** https://github.com/amirhossein-yousefi/ASR
- **Paper :** Baevski et al., “wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Representations” (arXiv:2006.11477)
- **Demo :** N/A (local CLI and SageMaker examples included)

## Uses

### Direct Use

- General‑purpose **English** speech transcription for short to moderate audio segments (default duration filter: ~1–18 seconds).
- Local batch transcription via CLI or Python, or real‑time deployment via AWS SageMaker (JSON base64 or raw WAV content types).

### Downstream Use 

- Domain adaptation / further fine‑tuning on task‑ or accent‑specific datasets.
- Export to **ONNX** for CPU‑friendly inference and integration in production applications.

### Out-of-Scope Use

- **Speaker diarization**, **punctuation restoration**, and **true streaming ASR** are not included.
- Multilingual or code‑switched speech without additional fine‑tuning.
- Very long files without chunking; heavy background noise without augmentation/tuning.

## Bias, Risks, and Limitations

- The default fine‑tuning dataset (**Common Voice 17.0, English**) can reflect collection biases (microphone quality, accents, demographics). Accuracy may degrade on out‑of‑domain audio (e.g., telephony, medical terms).
- Transcriptions may contain mistakes and can include sensitive/PII if present in audio; handle outputs responsibly.

### Recommendations

- Always evaluate **WER/CER** on your own hold‑out data. Consider adding punctuation casing models and domain vocabularies as needed.
- For regulated contexts, incorporate a human‑in‑the‑loop review and data governance.

## How to Get Started with the Model

**Python (local inference):**
```python
import torch, torchaudio
from transformers import AutoModelForCTC, AutoProcessor

model_dir = "./outputs/asr"  # or a Hugging Face hub id
device = "cuda" if torch.cuda.is_available() else "cpu"

processor = AutoProcessor.from_pretrained(model_dir)
model = AutoModelForCTC.from_pretrained(model_dir).to(device).eval()

wav, sr = torchaudio.load("path/to/file.wav")
target_sr = processor.feature_extractor.sampling_rate
if sr != target_sr:
    wav = torchaudio.functional.resample(wav, sr, target_sr)

inputs = processor(wav.squeeze(0).numpy(), sampling_rate=target_sr, return_tensors="pt", padding=True)
with torch.no_grad():
    logits = model(**{k: v.to(device) for k, v in inputs.items()}).logits
pred_ids = torch.argmax(logits, dim=-1)
print(processor.batch_decode(pred_ids.cpu().numpy())[0])
```

**CLI (example):**
```bash
python src/infer.py --model_dir ./outputs/asr --audio path/to/file.wav
```

## Training Details

### Training Data

- **Dataset:** Common Voice 17.0 (English), text column: `sentence`
- **Duration filter:** min ~1.0s, max ~18.0s
- **Notes:** Case‑aware normalization, whitelist filtering to match tokenizer vocabulary; optional waveform augmentations.

### Training Procedure

#### Preprocessing [optional]

- Robust audio decoding (FFmpeg preferred on Windows; fallback to `torchaudio/soundfile/librosa`), resampling to 16 kHz as required by Wav2Vec2.
- Tokenization via the model’s processor; dynamic padding with a **CTC** collator.

#### Training Hyperparameters

- **Epochs:** 3
- **Per‑device batch size:** 8 (× **8** grad accumulation → effective **64**)
- **Learning rate:** 3e‑5
- **Warmup ratio:** 0.05
- **Optimizer:** `adamw_torch_fused`
- **Weight decay:** 0.0
- **Precision:** FP16
- **Max grad norm:** 1.0
- **Logging:** every 50 steps; **Eval/Save:** every 500 steps; keep last 2 checkpoints; early stopping patience = 3
- **Seed:** 42

#### Speeds, Sizes, Times [optional]

- **Total FLOPs (training):** 10,814,747,992,293,114,000
- **Training runtime:** ~11,168 s for 2,346 steps
- **Logs:** TensorBoard at `src/output/logs` (or similar path as configured)

### Evaluation

#### Testing Data, Factors & Metrics

- **Metrics:** **WER** (primary) and **CER** (auxiliary), computed with `jiwer` utilities.
- **Factors:** English speech across CV17 splits; performance varies by accent, recording conditions, and utterance length.

#### Results

- Training includes **loss**, **eval WER**, and **eval CER** curves. See the `assets/` directory for plots.

#### Summary

- Baseline WER/CER are logged per‑eval; users should report domain‑specific results on their own datasets.

## Model Examination 

- Greedy decoding by default; beam search/LM fusion is not included in this repo. Inspect logits and alignments if needed for error analysis.

## Environmental Impact

- **Hardware Type:** Laptop (Windows)
- **GPU:** NVIDIA GeForce RTX 3080 Ti Laptop GPU (16 GB VRAM), Driver 576.52
- **CUDA / PyTorch:** CUDA 12.9, PyTorch 2.8.0+cu129
- **Hours used:** ~3.1 h (approx.)
- **Cloud Provider:** N/A for local; **AWS SageMaker** utilities available for cloud training/deployment
- **Compute Region:** N/A (local)
- **Carbon Emitted:** Not calculated; estimate with the [MLCO2 calculator](https://mlco2.github.io/impact#compute)

## Technical Specifications 

### Model Architecture and Objective

- **Architecture:** Wav2Vec2 encoder with **CTC** output layer
- **Objective:** Character‑level CTC loss for ASR

### Compute Infrastructure

#### Hardware

- Local GPU as above; or AWS instance types via SageMaker scripts (e.g., `ml.g4dn.xlarge`).

#### Software

- Python 3.10+
- Key dependencies: `transformers`, `datasets`, `torch`, `torchaudio`, `soundfile`, `librosa`, `jiwer`, `onnxruntime` (for ONNX testing), and `boto3`/`sagemaker` for deployment.

## Citation 

**BibTeX:**
```bibtex
@article{baevski2020wav2vec,
  title={wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Representations},
  author={Baevski, Alexei and Zhou, Henry and Mohamed, Abdelrahman and Auli, Michael},
  journal={arXiv preprint arXiv:2006.11477},
  year={2020}
}
```

**APA:**
Baevski, A., Zhou, H., Mohamed, A., & Auli, M. (2020). *wav2vec 2.0: A framework for self‑supervised learning of speech representations*. arXiv:2006.11477.

## Glossary 

- **WER**: Word Error Rate; lower is better.
- **CER**: Character Error Rate; lower is better.
- **CTC**: Connectionist Temporal Classification, an alignment‑free loss for sequence labeling.

## More Information 

- **ONNX export:** `src/export_onnx.py`
- **AWS SageMaker:** scripts in `sagemaker/` for training, deployment, and autoscaling.
- **Training/metrics plots:** see `assets/` (e.g., `train_loss.svg`, `eval_wer.svg`, `eval_cer.svg`).

## Model Card Authors 

- Amirhossein Yousefi (repo author)

## Model Card Contact

- Open an issue on the GitHub repository: https://github.com/amirhossein-yousefi/ASR