gbn2 / app.py
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Update app.py
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import soundfile as sf
import torch
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import gradio as gr
import sox
import subprocess
import openai
# Set your OpenAI API key
api_key = "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"
# spell_checker = GoogleSpellChecker(lang="fa")
def read_file_and_process(wav_file):
filename = wav_file.split('.')[0]
filename_16k = filename + "16k.wav"
resampler(wav_file, filename_16k)
speech, _ = sf.read(filename_16k)
inputs = processor(speech, sampling_rate=16_000, return_tensors="pt", padding=True)
return inputs
def resampler(input_file_path, output_file_path):
command = (
f"ffmpeg -hide_banner -loglevel panic -i {input_file_path} -ar 16000 -ac 1 -bits_per_raw_sample 16 -vn "
f"{output_file_path}"
)
subprocess.call(command, shell=True)
def parse_transcription(logits):
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.decode(predicted_ids[0], skip_special_tokens=True)
del(logits)
return transcription
# def corrector(sentence):
# check_spell = spell_checker.check(sentence)
# if check_spell[1] is None:
# return sentence
# else:
# return check_spell[1]
def correct_text_with_gpt(text):
openai.api_key = api_key
response = openai.Completion.create(
engine="text-davinci-003",
prompt=f"Please correct the following text: '{text}'\n\nCorrected text:",
max_tokens=1000,
temperature=0.5, # Temperature controls the randomness of the model's output. A higher value like 1.0 makes the output more random, while a lower value like 0.2 makes it more deterministic and focused.
top_p=1.0, # This parameter controls the diversity of the output. It sets a threshold for the cumulative probability of words to keep. Smaller values like 0.2 will result in more focused responses, while larger values like 0.8 will allow for more diversity.
frequency_penalty=0.2, # encourages the use of less common words
presence_penalty=0.5, # discourages the use of common words.
)
return response.choices[0].text.strip()
def parse(wav_file):
input_values = read_file_and_process(wav_file)
with torch.no_grad():
logits = model(**input_values).logits
return correct_text_with_gpt(parse_transcription(logits))
# def parse(wav_file):
# check_spell = ''
# input_values = read_file_and_process(wav_file)
# with torch.no_grad():
# logits = model(**input_values).logits
# # sentence = parse_transcription(logits)
# check_spell = spell_checker.check(parse_transcription(logits))
# # if check_spell[0] is False:
# # corrected = check_spell[1]
# # else:
# # corrected = sentence
# return spell_checker.check(parse_transcription(logits))[1] if spell_checker.check(parse_transcription(logits))[0] is False else parse_transcription(logits)
model_id = "jonatasgrosman/wav2vec2-large-xlsr-53-persian"
processor = Wav2Vec2Processor.from_pretrained(model_id)
model = Wav2Vec2ForCTC.from_pretrained(model_id)
input_ = gr.Audio(source="microphone",
type="filepath",
label="لطفا دکمه ضبط صدا را بزنید و شروع به صحبت کنید و بعذ از اتمام صحبت دوباره دکمه ضبط را فشار دهید.",
show_download_button=True,
show_edit_button=True,
)
txtbox = gr.Textbox(
label="متن گفتار شما: ",
lines=5,
text_align="right",
show_label=True,
show_copy_button=True,
)
title = "Speech-to-Text (persian)"
description = "، توجه داشته باشید که هرچه گفتار شما شمرده تر باشد خروجی با کیفیت تری دارید.روی دکمه ضبط صدا کلیک کنید و سپس دسترسی مرورگر خود را به میکروفون دستگاه بدهید، سپس شروع به صحبت کنید و برای اتمام ضبط دوباره روی دکمه کلیک کنید"
article = "<p style='text-align: center'><a href='https://github.com/nimaprgrmr'>Large-Scale Self- and Semi-Supervised Learning for Speech Translation</a></p>"
demo = gr.Interface(fn=parse, inputs = input_, outputs=txtbox, title=title, description=description, article = article,
streaming=True, interactive=True,
analytics_enabled=False, show_tips=False, enable_queue=True)
demo.launch(share=True)