Spaces:
Sleeping
Sleeping
jtlonsako
commited on
Commit
β’
a95e64c
1
Parent(s):
2a01fc3
Test 2
Browse files
app.py
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import soundfile as sf
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import datetime
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from pyctcdecode import BeamSearchDecoderCTC
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import torch
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import os
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import time
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import gc
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import gradio as gr
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import librosa
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from transformers import Wav2Vec2ForCTC, Wav2Vec2ProcessorWithLM, AutoModelForSeq2SeqLM, AutoTokenizer
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from GPUtil import showUtilization as gpu_usage
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from numba import cuda
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from google.cloud import translate
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# load pretrained model
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model = Wav2Vec2ForCTC.from_pretrained("facebook/mms-1b-all")
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processor = Wav2Vec2ProcessorWithLM.from_pretrained("jlonsako/mms-1b-all-AmhLM")
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#Define Functions
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#convert time into .sbv format
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def format_time(seconds):
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# Convert seconds to hh:mm:ss,ms format
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return str(datetime.timedelta(seconds=seconds)).replace('.', ',')
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#function to send text strings to be translated into english
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def translate_text(
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text: str = "αααα« α αα α΅αα
α¨ααα£α α₯α© α₯ααα₯ααα α₯αα΄α΅",
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project_id: str = "noble-feat-390914"
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) -> translate.TranslationServiceClient:
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"""Translating Text."""
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client = translate.TranslationServiceClient()
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location = "global"
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parent = f"projects/{project_id}/locations/{location}"
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# Translate text from English to Amharic
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# Detail on supported types can be found here:
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# https://cloud.google.com/translate/docs/supported-formats
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response = client.translate_text(
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request={
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"parent": parent,
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"contents": [text],
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"mime_type": "text/plain", # mime types: text/plain, text/html
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"source_language_code": "am",
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"target_language_code": "en-US",
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}
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)
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# Display the translation for each input text provided
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#for translation in response.translations:
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#print(f"Translated text: {translation.translated_text}")
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return response
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#Convert Video/Audio into 16K wav file
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def preprocessAudio(audioFile):
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os.system(f"ffmpeg -y -i {audioFile.name} -ar 16000 ./audio.wav")
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#Transcribe!!!
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def Transcribe(file):
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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start_time = time.time()
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model.load_adapter("amh")
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preprocessAudio(file)
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#os.system(f"ffmpeg -y -i ./July3_2023_Sermon.mov -ar 16000 ./audio.wav")
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block_size = 30
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transcripts = []
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stream = librosa.stream(
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"./audio.wav",
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block_length=block_size,
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frame_length=16000,
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hop_length=16000
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)
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model.to(device)
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print("Model loaded to gpu: Entering transcription phase")
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#Code for timestamping
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encoding_start = 0
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sbv_file = open("subtitle.sbv", "w")
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for speech_segment in stream:
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if len(speech_segment.shape) > 1:
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speech_segment = speech_segment[:,0] + speech_segment[:,1]
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input_values = processor(speech_segment, sampling_rate=16_000, return_tensors="pt").input_values.to(device)
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with torch.no_grad():
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logits = model(input_values).logits
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if len(logits.shape) == 1:
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print("test")
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logits = logits.unsqueeze(0)
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#predicted_ids = torch.argmax(logits, dim=-1)
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transcription = processor.batch_decode(logits.cpu().numpy()).text
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transcripts.append(transcription[0])
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#Generate timestamps
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encoding_end = encoding_start + block_size
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formatted_start = format_time(encoding_start)
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formatted_end = format_time(encoding_end)
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#Write to the .sbv file
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sbv_file.write(f"{formatted_start},{formatted_end}\n")
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sbv_file.write(f"{transcription[0]}\n\n")
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encoding_start = encoding_end
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# Freeing up memory
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del input_values
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del logits
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#del predicted_ids
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del transcription
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torch.cuda.empty_cache()
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gc.collect()
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# Join all transcripts into a single transcript
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transcript = ' '.join(transcripts)
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sbv_file.close()
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end_time = time.time()
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os.system("rm ./audio.wav")
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print(f"The script ran for {end_time - start_time} seconds.")
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return("subtitle.sbv")
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demo = gr.Interface(fn=Transcribe, inputs=gr.File(), outputs=gr.File())
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#with gr.Blocks() as demo:
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#file_output = gr.Textbox()
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#upload_button = gr.UploadButton("Click to Upload a sermon",
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# file_types=["video", "audio"], file_count="multiple")
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#upload_button.upload(Transcribe, upload_button, file_output)
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demo.launch()
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