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import gradio as gr
from transformers import Wav2Vec2ForCTC, AutoProcessor
import torch
import numpy as np
import librosa
model_id = "facebook/mms-1b-all"
def transcribe(audio_file_mic=None, audio_file_upload=None):
if audio_file_mic:
audio_file = audio_file_mic
elif audio_file_upload:
audio_file = audio_file_upload
else:
return "Please upload an audio file or record one"
speech, sample_rate = librosa.load(audio_file)
if sample_rate != 16000:
speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000)
processor = AutoProcessor.from_pretrained(model_id)
model = Wav2Vec2ForCTC.from_pretrained(model_id)
inputs = processor(speech, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
ids = torch.argmax(outputs, dim=-1)[0]
transcription = processor.decode(ids)
return transcription
iface = gr.Interface(fn=transcribe,
inputs=[
gr.Audio(source="microphone", type="filepath"),
gr.Audio(source="upload", type="filepath")
],
outputs=["textbox"],
)
iface.launch() |