wav2vec2-large-xlsr-53-punjabi

The kingabzpro/wav2vec2-large-xlsr-53-punjabi model is a Punjabi‑specific automatic‑speech‑recognition (ASR) system built on top of Facebook’s wav2vec 2.0 large XLSR‑53

kingabzpro 244K downloads apache-2.0 Speech Recognition
Frameworkstransformerspytorchsafetensors
Datasetsmozilla-foundation/common_voice_8_0
Tagstensorboardwav2vec2automatic-speech-recognitionhf-asr-leaderboardrobust-speech-eventpabase_model:Harveenchadha/vakyansh-wav2vec2-punjabi-pam-10base_model:finetune:Harveenchadha/vakyansh-wav2vec2-punjabi-pam-10
Downloads
244K
License
apache-2.0
Pipeline
Speech Recognition
Author
kingabzpro

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Technical Overview

The kingabzpro/wav2vec2-large-xlsr-53-punjabi model is a Punjabi‑specific automatic‑speech‑recognition (ASR) system built on top of Facebook’s wav2vec 2.0 large XLSR‑53 architecture. It takes raw audio waveforms, extracts high‑level acoustic representations with a deep transformer encoder, and decodes them into text using a Connectionist Temporal Classification (CTC) head. The model has been fine‑tuned on the Mozilla Common Voice pa‑IN dataset (version 8.0), making it one of the most robust Punjabi ASR solutions available on the Hugging Face Hub.

Key features and capabilities

  • Supports 48 kHz → 16 kHz resampling on‑the‑fly, enabling use with most consumer‑grade recordings.
  • Works with the AutoProcessor and AutoModelForCTC pipeline, simplifying integration in PyTorch, TensorFlow, or ONNX runtimes.
  • Compatible with language‑model (LM) rescoring for lower word‑error‑rate (WER) when a Punjabi LM is available.
  • Designed for robust speech‑event handling (background noise, reverberation) thanks to the XLSR‑53 pre‑training on 53 languages.

Architecture highlights

  • Base model: Harveenchadha/vakyansh-wav2vec2-punjabi-pam-10, itself a fine‑tuned wav2vec 2.0 large XLSR‑53 checkpoint.
  • Transformer encoder with 24 layers, 1024 hidden units, and 16 attention heads (≈300 M parameters).
  • CTC loss head trained on a character‑level Punjabi vocabulary (Unicode Gurmukhi script).
  • Native mixed‑precision (AMP) training for faster convergence on modern GPUs.

Intended use cases

  • Live transcription of Punjabi podcasts, news broadcasts, and webinars.
  • Voice‑controlled assistants and smart‑home devices targeting Punjabi‑speaking markets.
  • Automatic captioning for educational platforms and e‑learning content in Punjabi.
  • Batch processing of large audio archives for searchable text indexing.

Benchmark Performance

For ASR models, the most informative metrics are Word Error Rate (WER) and Character Error Rate (CER). Both measure the edit distance between the model’s transcription and a human‑generated reference, with lower values indicating higher accuracy.

Results on the Common Voice pa‑IN test split

  • WER (with language model): 36.02 %
  • CER (with language model): 12.81 %
  • WER (raw CTC output): 49.39 %
  • CER (raw CTC output): 22.38 %
  • Final validation loss: 1.2101

These numbers place the model among the top performers on the HF ASR Leaderboard for Punjabi, especially when an external language model is applied. The improvement from raw CTC to LM‑rescored results (≈13 % absolute WER reduction) demonstrates the model’s compatibility with downstream language‑model pipelines.

Hardware Requirements

Running a large wav2vec 2.0 model in real‑time requires a GPU with sufficient VRAM to hold the transformer weights and intermediate activations. The following guidelines are based on typical inference patterns for 16 kHz audio chunks of up to 30 seconds.

  • VRAM for inference: 8 GB (minimum) – 12 GB recommended for batch sizes ≥ 2.
  • Recommended GPU: NVIDIA RTX 3060 Ti, RTX 3070, or any GPU with ≥ 8 GB GDDR6 memory. For production‑scale deployments, consider RTX 3080/3090 or A100 for sub‑100 ms latency.
  • CPU: Modern multi‑core CPU (e.g., AMD Ryzen 5 5600X or Intel i7‑10700K) for preprocessing and audio resampling. CPU‑only inference is possible but will be 5‑10× slower.
  • Storage: The model checkpoint (~1.2 GB when stored as .safetensors) plus the tokenizer files (~200 KB). A fast SSD is recommended to avoid I/O bottlenecks.
  • Performance characteristics: On a RTX 3080, a 10‑second audio segment is transcribed in ~70 ms (including resampling). Latency scales linearly with audio length.

Use Cases

The model’s strong performance on Punjabi speech makes it a versatile building block across several domains.

  • Media & Entertainment: Automatic captioning for Punjabi TV channels, YouTube creators, and streaming services.
  • Education: Transcribing lecture recordings for e‑learning platforms, enabling searchable subtitles and note‑taking tools.
  • Customer Service: Speech‑to‑text for call‑center analytics, sentiment analysis, and real‑time agent assistance in Punjabi.
  • Accessibility: Voice‑controlled interfaces for visually impaired users, including smart‑home devices and mobile apps.
  • Legal & Government: Transcribing parliamentary sessions, public hearings, and court proceedings to improve transparency.

Integration is straightforward via the transformers library or ONNX Runtime, and the model can be combined with a Punjabi language model for higher accuracy in noisy environments.

Training Details

The model was fine‑tuned on the Mozilla Common Voice pa‑IN subset of version 8.0, using the Harveenchadha/vakyansh-wav2vec2-punjabi-pam-10 checkpoint as a starting point. Training was performed with the following hyper‑parameters:

  • Learning rate: 3e‑4 (linear scheduler with 200 warm‑up steps)
  • Batch size: 16 (effective batch size 32 due to gradient accumulation of 2)
  • Optimizer: Adam (β₁=0.9, β₂=0.999, ε=1e‑8)
  • Mixed‑precision (AMP) enabled for faster GPU utilization.
  • Number of epochs: 30 (early stopping after validation loss plateaued).
  • Random seed: 42 (ensures reproducibility).

Training was carried out on a single NVIDIA RTX 3090 (24 GB VRAM) using torch==1.10.2+cu102 and transformers==4.17.0.dev0. The loss curve shows a steady decline from 11.06 at epoch 3.7 to 0.2088 at epoch 29.6, with corresponding WER improvements from 71 % down to 49 % (raw) and 36 % after LM rescoring.

The model can be further fine‑tuned on domain‑specific Punjabi corpora (e.g., medical dictation) by loading the checkpoint with AutoModelForCTC.from_pretrained and continuing training with a lower learning rate (e.g., 1e‑5) to preserve the learned acoustic representations.

Licensing Information

The model card lists the Apache‑2.0 license, which is a permissive open‑source license. Although the top‑level metadata mentions an “unknown” license, the explicit license: apache-2.0 tag supersedes that and grants the following freedoms:

  • Free use for personal, academic, or commercial projects.
  • Permission to modify, distribute, and create derivative works.
  • Obligation to retain the original copyright notice and provide a copy of the Apache‑2.0 license in any redistributed version.
  • No warranty; the model is provided “as‑is”.

Because Apache‑2.0 is compatible with most proprietary software, you can embed this model in commercial products (e.g., voice assistants, transcription services) without needing to open‑source your own code. Just ensure you include the required attribution and license text in your documentation or “About” page.

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