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import streamlit as st | |
import os | |
import soundfile as sf | |
import torch | |
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline | |
device = "cuda:0" if torch.cuda.is_available() else "cpu" | |
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 | |
model_id = "distil-whisper/distil-large-v2" | |
model = AutoModelForSpeechSeq2Seq.from_pretrained( | |
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True | |
) | |
model.to(device) | |
processor = AutoProcessor.from_pretrained(model_id) | |
pipe = pipeline( | |
"automatic-speech-recognition", | |
model=model, | |
tokenizer=processor.tokenizer, | |
feature_extractor=processor.feature_extractor, | |
max_new_tokens=128, | |
chunk_length_s=15, | |
batch_size=16, | |
torch_dtype=torch_dtype, | |
device=device, | |
) | |
def transcribe_audio(audio_file): | |
# Save the audio file to a temporary file | |
with open("temp_audio_file", "wb") as f: | |
f.write(audio_file.getbuffer()) | |
# Transcribe the audio file using the Whisper model | |
result = pipe("temp_audio_file") | |
return result["text"] | |
# Streamlit app | |
def main(): | |
st.title('BETTER TRANSCRIBER') | |
# Audio file uploader | |
uploaded_file = st.file_uploader("Upload an audio file", type=["wav", "mp3", "m4a", "ogg", "flac"]) | |
if uploaded_file is not None: | |
# Show a button to start the transcription process | |
if st.button('Transcribe'): | |
# Show a message while transcribing | |
with st.spinner('Transcribing...'): | |
text = transcribe_audio(uploaded_file) | |
# Show the transcription | |
st.subheader('Transcription:') | |
st.write(text) | |
else: | |
st.write('Upload an audio file to get started.') | |
if __name__ == "__main__": | |
main() |