Amh-Transcribe / GradioApp.py
jtlonsako
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import soundfile as sf
import datetime
from pyctcdecode import BeamSearchDecoderCTC
import torch
import os
import time
import gc
import gradio as gr
import librosa
from transformers import Wav2Vec2ForCTC, Wav2Vec2ProcessorWithLM, AutoModelForSeq2SeqLM, AutoTokenizer
from GPUtil import showUtilization as gpu_usage
from numba import cuda
from google.cloud import translate
# load pretrained model
model = Wav2Vec2ForCTC.from_pretrained("facebook/mms-1b-all")
processor = Wav2Vec2ProcessorWithLM.from_pretrained("jlonsako/mms-1b-all-AmhLM")
#Define Functions
#convert time into .sbv format
def format_time(seconds):
# Convert seconds to hh:mm:ss,ms format
return str(datetime.timedelta(seconds=seconds)).replace('.', ',')
#function to send text strings to be translated into english
def translate_text(
text: str = "αˆƒαˆŒαˆ‰α‹« αŠ αˆαŠ• α‹΅αˆα… α‹¨αˆšα‹ˆαŒ£αˆ αŒ₯ሩ αŠ₯αŒα‹›α‰₯αˆ”αˆ­αŠ• αŠ₯αŠ•α‹΄α‰΅",
project_id: str = "noble-feat-390914"
) -> translate.TranslationServiceClient:
"""Translating Text."""
client = translate.TranslationServiceClient()
location = "global"
parent = f"projects/{project_id}/locations/{location}"
# Translate text from English to Amharic
# Detail on supported types can be found here:
# https://cloud.google.com/translate/docs/supported-formats
response = client.translate_text(
request={
"parent": parent,
"contents": [text],
"mime_type": "text/plain", # mime types: text/plain, text/html
"source_language_code": "am",
"target_language_code": "en-US",
}
)
# Display the translation for each input text provided
#for translation in response.translations:
#print(f"Translated text: {translation.translated_text}")
return response
#Convert Video/Audio into 16K wav file
def preprocessAudio(audioFile):
os.system(f"ffmpeg -y -i {audioFile.name} -ar 16000 ./audio.wav")
#Transcribe!!!
def Transcribe(file):
device = "cuda:0" if torch.cuda.is_available() else "cpu"
start_time = time.time()
model.load_adapter("amh")
preprocessAudio(file)
#os.system(f"ffmpeg -y -i ./July3_2023_Sermon.mov -ar 16000 ./audio.wav")
block_size = 30
transcripts = []
stream = librosa.stream(
"./audio.wav",
block_length=block_size,
frame_length=16000,
hop_length=16000
)
model.to(device)
print("Model loaded to gpu: Entering transcription phase")
#Code for timestamping
encoding_start = 0
sbv_file = open("subtitle.sbv", "w")
for speech_segment in stream:
if len(speech_segment.shape) > 1:
speech_segment = speech_segment[:,0] + speech_segment[:,1]
input_values = processor(speech_segment, sampling_rate=16_000, return_tensors="pt").input_values.to(device)
with torch.no_grad():
logits = model(input_values).logits
if len(logits.shape) == 1:
print("test")
logits = logits.unsqueeze(0)
#predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(logits.cpu().numpy()).text
transcripts.append(transcription[0])
#Generate timestamps
encoding_end = encoding_start + block_size
formatted_start = format_time(encoding_start)
formatted_end = format_time(encoding_end)
#Write to the .sbv file
sbv_file.write(f"{formatted_start},{formatted_end}\n")
sbv_file.write(f"{transcription[0]}\n\n")
encoding_start = encoding_end
# Freeing up memory
del input_values
del logits
#del predicted_ids
del transcription
torch.cuda.empty_cache()
gc.collect()
# Join all transcripts into a single transcript
transcript = ' '.join(transcripts)
sbv_file.close()
end_time = time.time()
os.system("rm ./audio.wav")
print(f"The script ran for {end_time - start_time} seconds.")
return("subtitle.sbv")
demo = gr.Interface(fn=Transcribe, inputs=gr.File(), outputs=gr.File())
#with gr.Blocks() as demo:
#file_output = gr.Textbox()
#upload_button = gr.UploadButton("Click to Upload a sermon",
# file_types=["video", "audio"], file_count="multiple")
#upload_button.upload(Transcribe, upload_button, file_output)
demo.launch()