Commit
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c106aba
1
Parent(s):
7c840cd
Create app.py
Browse files
app.py
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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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from glob import glob
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import torchaudio
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import subprocess
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import gradio as gr
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resampler = torchaudio.transforms.Resample(48_000, 16_000)
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def get_filename(wav_file):
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filename_local = wav_file.split('/')[-1][:-4]
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filename_new = '/tmp/'+filename_local+'_16.wav'
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subprocess.call(["sox {} -r {} -b 16 -c 1 {}".format(wav_file, str(16000), filename_new)], shell=True)
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return filename_new
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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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# load audio
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wav_file = get_filename(wav_file.name)
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audio_input, sample_rate = sf.read(wav_file)
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#test_file = resampler(test_file[0])
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# pad input values and return pt tensor
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input_values = processor(audio_input, sampling_rate=16_000, return_tensors="pt").input_values
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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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# transcribe
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transcription = processor.decode(predicted_ids[0], skip_special_tokens=True)
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return transcription
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input = gr.inputs.Audio(source="microphone", type="file")
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gr.Interface(parse_transcription, inputs = input, outputs="text",
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analytics_enabled=False, show_tips=False).launch(inline=False);
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