Automatic Speech Recognition
Transformers
Safetensors
Hindi
whisper
speech-recognition
hindi
hinglish
Instructions to use MmeOG/MME_STT_Model with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use MmeOG/MME_STT_Model with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="MmeOG/MME_STT_Model")# pip install -U transformers accelerate # Load model directly from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq processor = AutoProcessor.from_pretrained("MmeOG/MME_STT_Model") model = AutoModelForSpeechSeq2Seq.from_pretrained("MmeOG/MME_STT_Model", device_map="auto") - Notebooks
- Google Colab
- Kaggle
Add model card
Browse files
README.md
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license: other
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---
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---
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language:
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- hi
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library_name: transformers
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pipeline_tag: automatic-speech-recognition
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license: other
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tags:
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- whisper
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- speech-recognition
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- hindi
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- hinglish
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---
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# MME_STT_Model
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Hindi speech to Roman-script Hinglish transcription. This model was fine-tuned with LoRA and the adapter
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was merged into the weights, so it loads as a normal Transformers model with no PEFT.
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## Usage
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```python
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import torch, librosa
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from transformers import WhisperForConditionalGeneration, WhisperProcessor
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model = WhisperForConditionalGeneration.from_pretrained("MmeOG/MME_STT_Model", torch_dtype=torch.float16).to("cuda")
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processor = WhisperProcessor.from_pretrained("MmeOG/MME_STT_Model")
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audio, _ = librosa.load("sample.wav", sr=16000)
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feats = processor(audio, sampling_rate=16000, return_tensors="pt").input_features.to("cuda", torch.float16)
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ids = model.generate(feats, language="hi", task="transcribe")
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print(processor.batch_decode(ids, skip_special_tokens=True)[0])
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```
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## Training
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LoRA fine-tune on about 190k utterances: IndicVoices (about 92%), MUCS and UJS (about 4% each).
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Two epochs, batch size 16, learning rate 3e-5 (cosine), one A100 80GB.
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## Evaluation
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Phonetic WER (pWER, lower is better) on the public set `RinggAI/ASR-Benchmarking-Dataset`
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(sampled 200 clips per dataset, 1,100 clips in total). The merged model reproduces the adapter's published scores to within 0.05 points.
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| Dataset | This model | Before fine-tuning) |
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|---|---|---|
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| fleurs | 18.45 | 20.15 |
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| commonvoice | 26.42 | 29.16 |
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| indictts | 13.32 | 13.72 |
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| kathbath | 19.79 | 21.73 |
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| kathbath_noisy | 21.91 | 23.37 |
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| mucs | 19.65 | 20.84 |
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| **all** | **19.64** | **21.21** |
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pWER is a lenient metric that compares text in a shared phonetic form, so it ranks systems on the
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same clips but should not be read as an absolute accuracy figure.
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## Limitations
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- Not a streaming model: it transcribes after the speaker finishes.
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- Short utterances (under 2 seconds) have high error rates.
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- Accuracy drops sharply in loud background noise, especially babble.
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- Not evaluated on real phone-call recordings.
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- Commercial systems scored better on the same public benchmark.
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