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@@ -8,9 +8,11 @@ tags:
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  - audio
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  - automatic-speech-recognition
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  - speech
 
 
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  license: mit
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  model-index:
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- - name: Wav2Vec2 Vakyansh Hindi Model by Harveen Chadha
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  results:
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  - task:
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  name: Speech Recognition
@@ -23,116 +25,4 @@ model-index:
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  - name: Test WER
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  type: wer
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  value: 33.17
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- ---
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-
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- ## Spaces Demo
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- Check the spaces demo [here](https://huggingface.co/spaces/Harveenchadha/wav2vec2-vakyansh-hindi/tree/main)
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-
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- ## Pretrained Model
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-
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- Fine-tuned on Multilingual Pretrained Model [CLSRIL-23](https://arxiv.org/abs/2107.07402). The original fairseq checkpoint is present [here](https://github.com/Open-Speech-EkStep/vakyansh-models). When using this model, make sure that your speech input is sampled at 16kHz.
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-
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- **Note: The result from this model is without a language model so you may witness a higher WER in some cases.**
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-
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- ## Dataset
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-
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- This model was trained on 4200 hours of Hindi Labelled Data. The labelled data is not present in public domain as of now.
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-
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- ## Training Script
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-
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- Models were trained using experimental platform setup by Vakyansh team at Ekstep. Here is the [training repository](https://github.com/Open-Speech-EkStep/vakyansh-wav2vec2-experimentation).
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-
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- In case you want to explore training logs on wandb they are [here](https://wandb.ai/harveenchadha/hindi_finetuning_multilingual?workspace=user-harveenchadha).
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-
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-
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- ## [Colab Demo](https://colab.research.google.com/github/harveenchadha/bol/blob/main/demos/hf/hindi/hf_hindi_him_4200_demo.ipynb)
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-
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- ## Usage
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-
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- The model can be used directly (without a language model) as follows:
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-
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- ```python
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- import soundfile as sf
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- import torch
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- from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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- import argparse
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-
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- def parse_transcription(wav_file):
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- # load pretrained model
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- processor = Wav2Vec2Processor.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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- model = Wav2Vec2ForCTC.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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-
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- # load audio
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- audio_input, sample_rate = sf.read(wav_file)
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-
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- # pad input values and return pt tensor
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- input_values = processor(audio_input, sampling_rate=sample_rate, return_tensors="pt").input_values
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-
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- # INFERENCE
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- # retrieve logits & take argmax
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- logits = model(input_values).logits
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- predicted_ids = torch.argmax(logits, dim=-1)
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-
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- # transcribe
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- transcription = processor.decode(predicted_ids[0], skip_special_tokens=True)
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- print(transcription)
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-
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- ```
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-
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-
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- ## Evaluation
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- The model can be evaluated as follows on the hindi test data of Common Voice.
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-
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- ```python
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-
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- import torch
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- import torchaudio
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- from datasets import load_dataset, load_metric
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- from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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- import re
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-
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- test_dataset = load_dataset("common_voice", "hi", split="test")
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- wer = load_metric("wer")
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-
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- processor = Wav2Vec2Processor.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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- model = Wav2Vec2ForCTC.from_pretrained("Harveenchadha/vakyansh-wav2vec2-hindi-him-4200")
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- model.to("cuda")
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-
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- resampler = torchaudio.transforms.Resample(48_000, 16_000)
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-
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- chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\“]'
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-
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- # Preprocessing the datasets.
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- # We need to read the aduio files as arrays
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- def speech_file_to_array_fn(batch):
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- batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
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- speech_array, sampling_rate = torchaudio.load(batch["path"])
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- batch["speech"] = resampler(speech_array).squeeze().numpy()
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- return batch
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-
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- test_dataset = test_dataset.map(speech_file_to_array_fn)
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-
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- # Preprocessing the datasets.
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- # We need to read the aduio files as arrays
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- def evaluate(batch):
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- inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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-
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- with torch.no_grad():
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- logits = model(inputs.input_values.to("cuda")).logits
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-
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- pred_ids = torch.argmax(logits, dim=-1)
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- batch["pred_strings"] = processor.batch_decode(pred_ids, skip_special_tokens=True)
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- return batch
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- result = test_dataset.map(evaluate, batched=True, batch_size=8)
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- print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
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-
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- ```
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-
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- **Test Result**: 33.17 %
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- [**Colab Evaluation**](https://colab.research.google.com/github/harveenchadha/bol/blob/main/demos/hf/hindi/hf_vakyansh_hindi_him_4200_evaluation_common_voice.ipynb)
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-
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- ## Credits
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- Thanks to Ekstep Foundation for making this possible. The vakyansh team will be open sourcing speech models in all the Indic Languages.
 
8
  - audio
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  - automatic-speech-recognition
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  - speech
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+ - hindi asr
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+ - hindi speech recognition
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  license: mit
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  model-index:
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+ - name: Wav2Vec2 Vakyansh Hindi Model by Yuvraj Mor
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  results:
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  - task:
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  name: Speech Recognition
 
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  - name: Test WER
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  type: wer
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  value: 33.17
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+ ---