adnaniqbal001 commited on
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054bba0
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1 Parent(s): 92f1a16

Update app.py

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Files changed (1) hide show
  1. app.py +45 -31
app.py CHANGED
@@ -1,57 +1,71 @@
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-
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- # app.py
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  import streamlit as st
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  import torch
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- from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC, MarianMTModel, MarianTokenizer
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  import soundfile as sf
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  import tempfile
 
8
 
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  # Load models and tokenizers
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  @st.cache_resource
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  def load_models():
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- # Load ASR model (Wav2Vec2 for Urdu)
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- asr_processor = Wav2Vec2Processor.from_pretrained("m3hrdadfi/wav2vec2-large-xlsr-ur")
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- asr_model = Wav2Vec2ForCTC.from_pretrained("m3hrdadfi/wav2vec2-large-xlsr-ur")
 
 
 
 
 
 
 
 
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- # Load translation model (Urdu to German)
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- translation_tokenizer = MarianTokenizer.from_pretrained("Helsinki-NLP/opus-mt-ur-de")
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- translation_model = MarianMTModel.from_pretrained("Helsinki-NLP/opus-mt-ur-de")
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- return asr_processor, asr_model, translation_tokenizer, translation_model
 
 
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  asr_processor, asr_model, translation_tokenizer, translation_model = load_models()
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- # Streamlit App UI
 
 
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  st.title("Real-Time Urdu to German Voice Translator")
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- st.markdown("Upload an Urdu audio file, and the app will translate it to German.")
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- uploaded_file = st.file_uploader("Upload an audio file (in .wav format)", type=["wav"])
 
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  if uploaded_file is not None:
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  with tempfile.NamedTemporaryFile(delete=False) as temp_file:
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  temp_file.write(uploaded_file.read())
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  temp_file_path = temp_file.name
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- # Load audio file
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- audio_input, sample_rate = sf.read(temp_file_path)
 
 
 
 
 
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- # Ensure proper sampling rate
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- if sample_rate != 16000:
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- st.error("Please upload a .wav file with a sampling rate of 16kHz.")
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- else:
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- st.info("Processing the audio...")
 
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- # Convert speech to text (ASR)
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- input_values = asr_processor(audio_input, return_tensors="pt", sampling_rate=16000).input_values
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- with torch.no_grad():
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- logits = asr_model(input_values).logits
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- predicted_ids = torch.argmax(logits, dim=-1)
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- transcription = asr_processor.batch_decode(predicted_ids)[0]
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- st.text(f"Transcribed Urdu Text: {transcription}")
 
 
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- # Translate Urdu text to German
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- translated = translation_model.generate(**translation_tokenizer(transcription, return_tensors="pt", padding=True))
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- german_translation = translation_tokenizer.decode(translated[0], skip_special_tokens=True)
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- st.success(f"Translated German Text: {german_translation}")
 
 
 
 
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  import streamlit as st
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  import torch
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+ from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC, MarianMTModel, MarianTokenizer, Wav2Vec2CTCTokenizer
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  import soundfile as sf
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  import tempfile
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+ import numpy as np
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  # Load models and tokenizers
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  @st.cache_resource
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  def load_models():
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+ try:
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+ # Load Wav2Vec2 for ASR (Multilingual model for Urdu support)
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+ # Load the tokenizer directly using Wav2Vec2CTCTokenizer
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+ tokenizer = Wav2Vec2CTCTokenizer.from_pretrained("facebook/wav2vec2-large-xlsr-53")
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+ # Then, initialize the processor with the tokenizer
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+ asr_processor = Wav2Vec2Processor(feature_extractor=asr_processor.feature_extractor, tokenizer=tokenizer)
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+ asr_model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-xlsr-53")
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+
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+ # Load MarianMT for translation (Urdu to German)
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+ translation_tokenizer = MarianTokenizer.from_pretrained("Helsinki-NLP/opus-mt-ur-de")
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+ translation_model = MarianMTModel.from_pretrained("Helsinki-NLP/opus-mt-ur-de")
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+ return asr_processor, asr_model, translation_tokenizer, translation_model
 
 
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+ except Exception as e:
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+ st.error(f"Error loading models: {e}")
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+ return None, None, None, None
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+
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+ # Initialize models
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  asr_processor, asr_model, translation_tokenizer, translation_model = load_models()
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+ # ... (rest of your app.py code remains the same)
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+
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+ # Streamlit app interface
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  st.title("Real-Time Urdu to German Voice Translator")
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+ st.markdown("Upload an Urdu audio file in `.wav` format, and the app will transcribe and translate it.")
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+ # File uploader
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+ uploaded_file = st.file_uploader("Upload your Urdu audio file (16kHz .wav)", type=["wav"])
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  if uploaded_file is not None:
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  with tempfile.NamedTemporaryFile(delete=False) as temp_file:
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  temp_file.write(uploaded_file.read())
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  temp_file_path = temp_file.name
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+ try:
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+ # Load and validate audio file
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+ audio_input, sample_rate = sf.read(temp_file_path)
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+ if sample_rate != 16000:
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+ st.error("Audio file must have a sampling rate of 16kHz.")
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+ else:
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+ st.info("Processing the audio...")
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+ # Step 1: Speech-to-Text (ASR)
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+ input_values = asr_processor(audio_input, return_tensors="pt", sampling_rate=16000).input_values
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+ with torch.no_grad():
58
+ logits = asr_model(input_values).logits
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+ predicted_ids = torch.argmax(logits, dim=-1)
60
+ transcription = asr_processor.batch_decode(predicted_ids)[0]
61
 
62
+ st.text(f"Transcribed Urdu Text: {transcription}")
 
 
 
 
 
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64
+ # Step 2: Translate Text (Urdu to German)
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+ translated = translation_model.generate(**translation_tokenizer(transcription, return_tensors="pt", padding=True))
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+ german_translation = translation_tokenizer.decode(translated[0], skip_special_tokens=True)
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+ st.success(f"Translated German Text: {german_translation}")
 
 
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70
+ except Exception as e:
71
+ st.error(f"An error occurred: {e}")