Faster-Whisper / app.py
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Update app.py
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from transformers import (
AutomaticSpeechRecognitionPipeline,
WhisperForConditionalGeneration,
WhisperTokenizer,
WhisperProcessor,
)
from peft import PeftModel, PeftConfig
import torch
from huggingface_hub import snapshot_download, login
login()
peft_model_id = "aisha-org/faster-whisper-uz"
language = "uz"
task = "transcribe"
peft_config = PeftConfig.from_pretrained(peft_model_id, use_auth_token=True)
model = WhisperForConditionalGeneration.from_pretrained(
peft_config.base_model_name_or_path,
load_in_8bit=True,
device_map="auto",
use_auth_token=True,
force_download=True,
resume_download=False
)
model = PeftModel.from_pretrained(model, peft_model_id, use_auth_token=True)
tokenizer = WhisperTokenizer.from_pretrained(peft_config.base_model_name_or_path, language=language, task=task)
processor = WhisperProcessor.from_pretrained(peft_config.base_model_name_or_path, language=language, task=task)
feature_extractor = processor.feature_extractor
forced_decoder_ids = processor.get_decoder_prompt_ids(language=language, task=task)
pipe = AutomaticSpeechRecognitionPipeline(model=model, tokenizer=tokenizer, feature_extractor=feature_extractor)
def transcribe(audio):
with torch.cuda.amp.autocast():
text = pipe(audio, generate_kwargs={"forced_decoder_ids": forced_decoder_ids}, max_new_tokens=255)["text"]
return text
import gradio as gr
demo = gr.Blocks()
mic_transcribe = gr.Interface(
fn=transcribe,
inputs=gr.Audio(sources="microphone", type="filepath"),
outputs=gr.Textbox(),
)
file_transcribe = gr.Interface(
fn=transcribe,
inputs=gr.Audio(sources="upload", type="filepath"),
outputs=gr.Textbox(),
)
with demo:
gr.TabbedInterface(
[mic_transcribe, file_transcribe],
["Transcribe Microphone", "Transcribe Audio File"],
)
demo.launch(debug=True)