import warnings warnings.filterwarnings("ignore") import os import re import librosa import webrtcvad import nbimporter import torchaudio import numpy as np import gradio as gr import scipy.signal import soundfile as sf from transformers import pipeline from transformers import AutoProcessor from pyctcdecode import build_ctcdecoder from transformers import Wav2Vec2ProcessorWithLM from text2int import text_to_int from isNumber import is_number from Text2List import text_to_list from convert2list import convert_to_list from processDoubles import process_doubles from replaceWords import replace_words from applyVad import apply_vad from wienerFilter import wiener_filter from highPassFilter import high_pass_filter from wavletDenoise import wavelet_denoise transcriber_hindi_new = pipeline(task="automatic-speech-recognition", model="cdactvm/w2v-bert-2.0-hindi_new") transcriber_hindi_old = pipeline(task="automatic-speech-recognition", model="cdactvm/huggingface-hindi_model") processor = AutoProcessor.from_pretrained("cdactvm/w2v-bert-2.0-hindi_new") vocab_dict = processor.tokenizer.get_vocab() sorted_vocab_dict = {k.lower(): v for k, v in sorted(vocab_dict.items(), key=lambda item: item[1])} decoder = build_ctcdecoder( labels=list(sorted_vocab_dict.keys()), kenlm_model_path="lm.binary", ) processor_with_lm = Wav2Vec2ProcessorWithLM( feature_extractor=processor.feature_extractor, tokenizer=processor.tokenizer, decoder=decoder ) processor.feature_extractor._processor_class = "Wav2Vec2ProcessorWithLM" transcriber_hindi_lm = pipeline("automatic-speech-recognition", model="cdactvm/w2v-bert-2.0-hindi_new", tokenizer=processor_with_lm, feature_extractor=processor_with_lm.feature_extractor, decoder=processor_with_lm.decoder) def transcribe_hindi_new(audio): # # Process the audio file transcript = transcriber_hindi_new(audio) text_value = transcript['text'] processd_doubles=process_doubles(text_value) replaced_words = replace_words(processd_doubles) converted_text=text_to_int(replaced_words) return converted_text def transcribe_hindi_lm(audio): # # Process the audio file transcript = transcriber_hindi_lm(audio) text_value = transcript['text'] processd_doubles=process_doubles(text_value) replaced_words = replace_words(processd_doubles) converted_text=text_to_int(replaced_words) return converted_text def transcribe_hindi_old(audio): # # Process the audio file transcript = transcriber_hindi_old(audio) text_value = transcript['text'] cleaned_text=text_value.replace("","") processd_doubles=process_doubles(cleaned_text) replaced_words = replace_words(processd_doubles) converted_text=text_to_int(replaced_words) return converted_text ############################################### # implementation of noise reduction techniques. # Function to apply a Wiener filter for noise reduction def apply_wiener_filter(audio): return wiener(audio) # Function to handle speech recognition def Noise_cancellation_function(audio_file): # Load the audio file using librosa audio, sr = librosa.load(audio_file, sr=16000) # Step 1: Apply a high-pass filter audio = high_pass_filter(audio, sr) # Step 2: Apply Wiener filter for noise reduction audio = apply_wiener_filter(audio) # Step 3: Apply wavelet denoising denoised_audio = wavelet_denoise(audio) # Save the denoised audio to a temporary file temp_wav = "temp_denoised.wav" write(temp_wav, sr, denoised_audio) # Perform speech recognition on the denoised audio transcript = transcriber_hindi_lm(temp_wav) text_value = transcript['text'] cleaned_text=text_value.replace("","") processd_doubles=process_doubles(cleaned_text) replaced_words = replace_words(processd_doubles) converted_text=text_to_int(replaced_words) return converted_text ################################################# def sel_lng(lng, mic=None, file=None): if mic is not None: audio = mic elif file is not None: audio = file else: return "You must either provide a mic recording or a file" if lng == "model_1": return transcribe_hindi_old(audio) elif lng == "model_2": return transcribe_hindi_new(audio) elif lng== "model_3": return transcribe_hindi_lm(audio) elif lng== "model_4": return Noise_cancellation_function(audio) # demo=gr.Interface( # transcribe, # inputs=[ # gr.Audio(sources=["microphone","upload"], type="filepath"), # ], # outputs=[ # "textbox" # ], # title="Automatic Speech Recognition", # description = "Demo for Automatic Speech Recognition. Use microphone to record speech. Please press Record button. Initially it will take some time to load the model. The recognized text will appear in the output textbox", # ).launch() demo=gr.Interface( fn=sel_lng, inputs=[ gr.Dropdown([ "model_1","model_2","model_3","model_4"],label="Select Model"), gr.Audio(sources=["microphone","upload"], type="filepath"), ], outputs=[ "textbox" ], title="Automatic Speech Recognition", description = "Demo for Automatic Speech Recognition. Use microphone to record speech. Please press Record button. Initially it will take some time to load the model. The recognized text will appear in the output textbox", ).launch()