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import soundfile as sf |
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import torch |
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor,Wav2Vec2ProcessorWithLM |
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import gradio as gr |
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import sox |
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import subprocess |
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def read_file_and_process(wav_file): |
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filename = wav_file.split('.')[0] |
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filename_16k = filename + "16k.wav" |
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resampler(wav_file, filename_16k) |
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speech, _ = sf.read(filename_16k) |
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inputs = processor(speech, sampling_rate=16_000, return_tensors="pt", padding=True) |
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return inputs |
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def resampler(input_file_path, output_file_path): |
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command = ( |
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f"ffmpeg -hide_banner -loglevel panic -i {input_file_path} -ar 16000 -ac 1 -bits_per_raw_sample 16 -vn " |
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f"{output_file_path}" |
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) |
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subprocess.call(command, shell=True) |
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def parse_transcription_with_lm(logits): |
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result = processor_with_LM.batch_decode(logits.cpu().numpy()) |
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text = result.text |
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transcription = text[0].replace('<s>','') |
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return transcription |
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def parse_transcription(logits): |
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predicted_ids = torch.argmax(logits, dim=-1) |
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transcription = processor.decode(predicted_ids[0], skip_special_tokens=True) |
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return transcription |
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def parse(wav_file, applyLM): |
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input_values = read_file_and_process(wav_file) |
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with torch.no_grad(): |
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logits = model(**input_values).logits |
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if applyLM: |
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return parse_transcription_with_lm(logits) |
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else: |
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return parse_transcription(logits) |
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model_id = "aditii09/hindi-asr" |
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processor = Wav2Vec2Processor.from_pretrained(model_id) |
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processor_with_LM = Wav2Vec2ProcessorWithLM.from_pretrained(model_id) |
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model = Wav2Vec2ForCTC.from_pretrained(model_id) |
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input_ = gr.Audio(source="microphone", type="filepath") |
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txtbox = gr.Textbox( |
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label="Output from model will appear here:", |
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lines=5 |
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) |
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chkbox = gr.Checkbox(label="Apply LM", value=False) |
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gr.Interface(parse, inputs = [input_, chkbox], outputs=txtbox, |
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streaming=True, interactive=True, |
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analytics_enabled=False, show_tips=False, enable_queue=True).launch(inline=False); |