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# Transform an audio to text script with language detection. | |
# Author: Pratiksha Patel | |
# Description: This script record the audio, transform it to text, detect the language of the file and save it to a txt file. | |
# import required modules | |
#import torch | |
#import streamlit as st | |
#from audio_recorder_streamlit import audio_recorder | |
#import numpy as np | |
#from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq | |
#def transcribe_audio(audio_bytes): | |
# processor = AutoProcessor.from_pretrained("openai/whisper-large") | |
# model = AutoModelForSpeechSeq2Seq.from_pretrained("openai/whisper-large") | |
# Convert audio bytes to numpy array | |
# audio_array = np.frombuffer(audio_bytes, dtype=np.int16) | |
# Normalize audio array | |
#audio_tensor = torch.tensor(audio_array, dtype=torch.float64) / 32768.0 | |
# Provide inputs to the processor | |
##inputs = processor(audio=audio_tensor, sampling_rate=16000, return_tensors="pt") | |
#input_features = processor(audio_tensor, sampling_rate=16000, return_tensors="pt").input_features | |
# generate token ids | |
#predicted_ids = model.generate(input_features) | |
# decode token ids to text | |
#transcription = processor.batch_decode(predicted_ids, skip_special_tokens=False) | |
#transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True) | |
#return transcription | |
# Streamlit app | |
#st.title("Audio to Text Transcription..") | |
#audio_bytes = audio_recorder(pause_threshold=3.0, sample_rate=16_000) | |
#if audio_bytes: | |
# st.audio(audio_bytes, format="audio/wav") | |
# transcription = transcribe_audio(audio_bytes) | |
# if transcription: | |
# st.write("Transcription:") | |
# st.write(transcription) | |
#else: | |
# st.write("Error: Failed to transcribe audio.") | |
#else: | |
# st.write("No audio recorded.") | |
import torch | |
import streamlit as st | |
from audio_recorder_streamlit import audio_recorder | |
import numpy as np | |
#from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq | |
# Load model directly | |
from transformers import AutoProcessor, AutoModelForPreTraining | |
def transcribe_audio(audio_bytes): | |
# processor = AutoProcessor.from_pretrained("facebook/s2t-small-librispeech-asr") | |
# model = AutoModelForSpeechSeq2Seq.from_pretrained("facebook/s2t-small-librispeech-asr") | |
processor = AutoProcessor.from_pretrained("facebook/wav2vec2-base") | |
model = AutoModelForPreTraining.from_pretrained("facebook/wav2vec2-base") | |
# Convert audio bytes to numpy array | |
audio_array = np.frombuffer(audio_bytes, dtype=np.int16) | |
# Normalize audio array | |
audio_tensor = torch.tensor(audio_array, dtype=torch.float64) / 32768.0 | |
# Provide inputs to the processor | |
input_features = processor(audio_tensor, sampling_rate=16000, return_tensors="pt").input_features | |
# generate token ids | |
predicted_ids = model.generate(input_features) | |
# decode token ids to text | |
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=False) | |
return transcription | |
# Streamlit app | |
st.title("Audio to Text Transcription..") | |
audio_bytes = audio_recorder(pause_threshold=3.0, sample_rate=16_000) | |
if audio_bytes: | |
st.audio(audio_bytes, format="audio/wav") | |
transcription = transcribe_audio(audio_bytes) | |
if transcription: | |
st.write("Transcription:") | |
st.write(transcription) | |
else: | |
st.write("Error: Failed to transcribe audio.") | |
else: | |
st.write("No audio recorded.") |