app.py

import gradio as gr
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
import torch
import phonemizer
import librosa
import base64


def lark(audioAsB64):
    # convert b64 audio to wav
    with open("audio.wav", "wb") as preWaveform:
        preWaveform.write(base64.b64encode())

    # processing
    processor = Wav2Vec2Processor.from_pretrained(
        "facebook/wav2vec2-xlsr-53-espeak-cv-ft"
    )
    model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-xlsr-53-espeak-cv-ft")

    waveform, sample_rate = librosa.load(
        "harvard.wav", sr=16000
    )  # Downsample 44.1kHz to 8kHz

    input_values = processor(
        waveform, sampling_rate=sample_rate, return_tensors="pt"
    ).input_values

    with torch.no_grad():
        logits = model(input_values).logits

    predicted_ids = torch.argmax(logits, dim=-1)
    transcription = processor.batch_decode(predicted_ids)

    return transcription


iface = gr.Interface(fn=lark, inputs="text", outputs="text")
iface.launch()