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import gradio as gr | |
import soundfile as sf | |
import torch | |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor, pipeline | |
MODEL_NAME = "mikr/w2v-bert-2.0-czech-colab-cv16" | |
lang = "cs" | |
device = 0 if torch.cuda.is_available() else "cpu" | |
model = Wav2Vec2ForCTC.from_pretrained(MODEL_NAME).to(device) | |
processor = Wav2Vec2Processor.from_pretrained(MODEL_NAME) | |
pipe = pipeline( | |
model=MODEL_NAME, | |
) | |
def transcribe(file_upload): | |
warn_output = "" | |
if (file_upload is None): | |
return "ERROR: You have to either use the microphone or upload an audio file" | |
file = file_upload | |
text = pipe(file)["text"] | |
return warn_output + text | |
def readwav(a_f): | |
wav, sr = sf.read(a_f, dtype=np.float32) | |
if len(wav.shape) == 2: | |
wav = wav.mean(1) | |
if sr != 16000: | |
wlen = int(wav.shape[0] / sr * 16000) | |
wav = signal.resample(wav, wlen) | |
return wav | |
def transcribe2(file_upload): | |
wav = readwav(file_upload) | |
with torch.inference_mode(): | |
input_values = processor(wav, sampling_rate=16000).input_values[0] | |
input_values = torch.tensor(input_values, device=device).unsqueeze(0) | |
logits = model(input_values).logits | |
pred_ids = torch.argmax(logits, dim=-1) | |
xcp = processor.batch_decode(pred_ids) | |
return xcp[0] | |
iface = gr.Interface( | |
fn=transcribe2, | |
inputs=[ | |
gr.File(type="binary", label="Upload Audio File"), # Audio file upload | |
], | |
outputs="text", | |
theme="huggingface", | |
title="Wav2Vec2-Bert demo - transcribe Czech Audio", | |
description=( | |
"Transcribe long-form microphone or audio inputs with the click of a button! Demo uses the fine-tuned" | |
f" checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) from Whisper Fine Tuning Sprint Event 2022 " | |
"and 🤗 Transformers to transcribe audio files of arbitrary length." | |
), | |
allow_flagging="never", | |
) | |
iface.launch() | |