abidanoaman/urdu-asr-distilled-base-enhanced
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Enhanced Distilled Urdu ASR - wav2vec2-base
State-of-the-art lightweight Urdu ASR model using advanced knowledge distillation techniques.
๐ฏ Model Overview
This model uses cutting-edge distillation techniques:
- โ Feature-level distillation: Matches intermediate layer representations
- โ Temperature scheduling: Adaptive softening of targets (4.0 โ 1.5)
- โ SpecAugment: Time/frequency masking for robustness
- โ Multi-loss optimization: Logits + CTC + Features
Performance
- โ Speed: 2.4x faster inference
- โ Size: 3.3x smaller (94M vs 315M parameters)
- โ Accuracy: Only 11.3% WER degradation
๐ Detailed Results
Student Model (This Model)
- WER: 49.40%
- CER: 19.38%
- Parameters: 94,417,083 (94M)
- Inference Speed: 0.010s/batch
Teacher Model (Original)
- WER: 38.06%
- CER: 14.75%
- Parameters: 315,499,195 (315M)
- Inference Speed: 0.023s/batch
Improvements vs Standard Distillation
- Better WER retention through feature matching
- More robust via SpecAugment regularization
- Smoother training via temperature scheduling
๐ฌ Technical Details
Distillation Architecture
Teacher (Large) Student (Base)
โ โ
โโ Layer 0 โโโโโโโโโโโโ> Layer 0 (Feature Match)
โโ Layer 6 โโโโโโโโโโโโ> Layer 3 (Feature Match)
โโ Layer 12 โโโโโโโโโโโ> Layer 6 (Feature Match)
โโ Layer 18 โโโโโโโโโโโ> Layer 9 (Feature Match)
โโ Layer 24 โโโโโโโโโโโ> Layer 12 (Feature Match)
โ โ
โโโโ Logits KL-Div โโโLoss Function
L = ฮฑยทKL(S||T) + ฮฒยทCTC(S,y) + ฮณยทMSE(H_s,H_t)
Where:
ฮฑ = 0.4 (logit distillation weight)
ฮฒ = 0.4 (hard CTC weight)
ฮณ = 0.2 (feature distillation weight)
T = temperature (scheduled 4.0โ1.5)SpecAugment Configuration
- Time masking: 2 masks ร 80 frames
- Frequency masking: 2 masks ร 27 bins
๐ป Usage
Basic Inference
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
import torch
import torchaudio
# Load model
processor = Wav2Vec2Processor.from_pretrained("abidanoaman/urdu-asr-distilled-base-enhanced")
model = Wav2Vec2ForCTC.from_pretrained("abidanoaman/urdu-asr-distilled-base-enhanced")
# Load audio
audio, sr = torchaudio.load("audio.wav")
if sr != 16000:
resampler = torchaudio.transforms.Resample(sr, 16000)
audio = resampler(audio)
# Transcribe
inputs = processor(audio.squeeze().numpy(), sampling_rate=16000, return_tensors="pt")
with torch.no_grad():
logits = model(inputs.input_values).logits
pred_ids = torch.argmax(logits, dim=-1)
transcription = processor.decode(pred_ids[0])
print(transcription)๐ Deployment
Perfect for production environments requiring:
- โ Real-time transcription (< 0.01s latency)
- โ Low memory footprint (94MB)
- โ Edge deployment (Raspberry Pi, mobile)
- โ Cost-efficient scaling
๐ Training Details
Enhanced Techniques
- Feature-level distillation: Match hidden representations across 5 layer pairs
- Temperature scheduling: Linear decay from 4.0 to 1.5
- SpecAugment: Robust to time/frequency variations
Hyperparameters
- Epochs: 50
- Learning rate: 3e-05
- Batch size: 4
- Loss weights: ฮฑ=0.4, ฮฒ=0.4, ฮณ=0.2
๐ฏ Benchmark Comparison
๐ Citation
@misc{urdu-asr-enhanced-2024,
author = {Abid Anoaman},
title = {Enhanced Distilled Urdu ASR with Feature Matching and SpecAugment},
year = {2024},
publisher = {HuggingFace},
url = {https://huggingface.co/abidanoaman/urdu-asr-distilled-base-enhanced}
}๐ License
Apache 2.0
๐ Acknowledgments
- Teacher Model: abidanoaman/urdu-asr-complete-ablation
- Base Architecture: facebook/wav2vec2-base
- Techniques: Feature Distillation, Temperature Scheduling, SpecAugment
