WhisperATC / app.py
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#%%
from huggingface_hub import login
from transformers import pipeline
import gradio as gr
import os
login(token=os.environ['hf_token'])
## Try to load a local model if available
# try:
# whisper = pipeline(model='/mnt/projects/whisper/WhisperANSP/Models/whisper-large-v2-atco2-asr-atcosim-ANSP-3h1m', task='automatic-speech-recognition')
# ttl = 'Whisper Large v2 - ATCO2-ATCOSIM-ANSP'
# dis = 'This demo will transcribe ATC audio files by using the Whisper Large v2 model fine-tuned on the ATCO2, ATCOSIM and ANSP datasets. \n \n Further it uses a Named Entity Recognition model to extract callsigns, commands and values from the transcription. \n This model is based on Google\'s BERT model and fine-tuned on the ATCO2 dataset.'
# except:
# whisper = pipeline(model='jlvdoorn/whisper-large-v2-atco2-asr-atcosim')
# ttl = 'Whisper Large v2 - ATCO2-ATCOSIM'
# dis = 'This demo will transcribe ATC audio files by using the Whisper Large v2 model fine-tuned on the ATCO2 and ATCOSIM datasets. \n \n Further it uses a Named Entity Recognition model to extract callsigns, commands and values from the transcription. \n This model is based on Google\'s BERT model and fine-tuned on the ATCO2 dataset.'
bert_atco_ner = pipeline(model='Jzuluaga/bert-base-ner-atc-en-atco2-1h')
whisper_v2 = pipeline(model='jlvdoorn/whisper-large-v2-atco2-asr-atcosim')
whisper_v3 = pipeline(model='jlvdoorn/whisper-large-v3-atco2-asr-atcosim')
#%%
def transcribe(audio_file, audio_mic, model_version):
if model_version == 'large-v2':
whisper = whisper_v2
ttl = 'Whisper Large v2 - ATCO2-ATCOSIM'
dis = 'This demo will transcribe ATC audio files by using the Whisper Large v2 model fine-tuned on the ATCO2 and ATCOSIM datasets. \n \n Further it uses a Named Entity Recognition model to extract callsigns, commands and values from the transcription. \n This model is based on Google\'s BERT model and fine-tuned on the ATCO2 dataset.'
elif model_version == 'large-v3':
whisper = whisper_v3
ttl = 'Whisper Large v3 - ATCO2-ATCOSIM'
dis = 'This demo will transcribe ATC audio files by using the Whisper Large v3 model fine-tuned on the ATCO2 and ATCOSIM datasets. \n \n Further it uses a Named Entity Recognition model to extract callsigns, commands and values from the transcription. \n This model is based on Google\'s BERT model and fine-tuned on the ATCO2 dataset.'
if audio_mic is not None:
return whisper(audio_mic)['text']
elif audio_file is not None:
return whisper(audio_file)['text']
else:
return 'There was no audio to transcribe...'
#%%
def extractCallSignCommand(transcription):
if type(transcription) is str:
result = bert_atco_ner(transcription)
callsigns = []
commands = []
values = []
for item in result:
if 'callsign' in item['entity']:
callsigns.append(item['word'])
if 'command' in item['entity']:
commands.append(item['word'])
if 'value' in item['entity']:
values.append(item['word'])
return 'Callsigns: ' + ', '.join(callsigns) + '\nCommands: ' + ', '.join(commands) + '\nValues: ' + ', '.join(values)
else:
return 'There was no transcription to extract a callsign or command from...'
#%%
def transcribeAndExtract(audio_file, audio_mic, transcribe_only, model_version):
transcription = transcribe(audio_file, audio_mic, model_version)
if not transcribe_only:
callSignCommandValues = extractCallSignCommand(transcription)
else:
callSignCommandValues = ''
return transcription, callSignCommandValues
#%%
iface = gr.Interface(
fn=transcribeAndExtract,
inputs=[
gr.Audio(source='upload', type='filepath', interactive=True),
gr.Audio(source='microphone', type='filepath'),
gr.Checkbox(label='Transcribe only', default=False),
gr.Dropdown(choices=['large-v2', 'large-v3'], value='large-v3', label='Whisper model version'),
],
outputs=[gr.Text(label='Transcription'), gr.Text(label='Callsigns, commands and values')],
title='Whisper',
description='Transcribe and extract',
)
#%%
#iface.launch(server_name='0.0.0.0', server_port=9000)
iface.launch()