Michael Hu commited on
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initial check in

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Files changed (6) hide show
  1. LICENSE +21 -0
  2. app.py +118 -0
  3. pyproject.toml +60 -0
  4. utils/stt.py +51 -0
  5. utils/translation.py +45 -0
  6. utils/tts.py +46 -0
LICENSE ADDED
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+ MIT License
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+
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+ Copyright (c) 2025 Michael
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+
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+ Permission is hereby granted, free of charge, to any person obtaining a copy
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+ of this software and associated documentation files (the "Software"), to deal
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+ in the Software without restriction, including without limitation the rights
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+ to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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+ copies of the Software, and to permit persons to whom the Software is
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+ furnished to do so, subject to the following conditions:
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+
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+ The above copyright notice and this permission notice shall be included in all
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+ copies or substantial portions of the Software.
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+
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+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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+ IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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+ FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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+ AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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+ LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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+ OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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+ SOFTWARE.
app.py ADDED
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+ """
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+ Main entry point for the Audio Translation Web Application
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+ Handles file upload, processing pipeline, and UI rendering
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+ """
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+
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+ import streamlit as st
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+ import os
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+ import time
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+ from dotenv import load_dotenv
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+ from utils.stt import transcribe_audio
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+ from utils.translation import translate_text
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+ from utils.tts import generate_speech
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+
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+ # Initialize environment configurations
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+ load_dotenv()
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+ os.makedirs("temp/uploads", exist_ok=True)
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+ os.makedirs("temp/outputs", exist_ok=True)
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+
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+ def configure_page():
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+ """Set up Streamlit page configuration"""
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+ st.set_page_config(
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+ page_title="Audio Translator",
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+ page_icon="🎧",
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+ layout="wide",
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+ initial_sidebar_state="expanded"
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+ )
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+ st.markdown("""
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+ <style>
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+ .reportview-container {margin-top: -2em;}
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+ #MainMenu {visibility: hidden;}
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+ .stDeployButton {display:none;}
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+ </style>
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+ """, unsafe_allow_html=True)
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+
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+ def handle_file_processing(upload_path):
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+ """
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+ Execute the complete processing pipeline:
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+ 1. Speech-to-Text (STT)
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+ 2. Machine Translation
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+ 3. Text-to-Speech (TTS)
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+ """
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+ progress_bar = st.progress(0)
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+ status_text = st.empty()
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+
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+ try:
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+ # STT Phase
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+ status_text.markdown("🔍 **Performing Speech Recognition...**")
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+ english_text = transcribe_audio(upload_path)
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+ progress_bar.progress(30)
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+
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+ # Translation Phase
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+ status_text.markdown("🌐 **Translating Content...**")
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+ chinese_text = translate_text(english_text)
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+ progress_bar.progress(60)
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+
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+ # TTS Phase
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+ status_text.markdown("🎵 **Generating Chinese Speech...**")
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+ output_path = generate_speech(chinese_text)
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+ progress_bar.progress(100)
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+
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+ # Display results
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+ status_text.success("✅ Processing Complete!")
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+ return english_text, chinese_text, output_path
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+
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+ except Exception as e:
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+ status_text.error(f"❌ Processing Failed: {str(e)}")
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+ st.exception(e)
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+ raise
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+
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+ def render_results(english_text, chinese_text, output_path):
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+ """Display processing results in organized columns"""
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+ st.divider()
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+
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+ col1, col2 = st.columns([2, 1])
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+ with col1:
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+ st.subheader("Recognition Results")
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+ st.code(english_text, language="text")
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+
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+ st.subheader("Translation Results")
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+ st.code(chinese_text, language="text")
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+
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+ with col2:
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+ st.subheader("Audio Output")
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+ st.audio(output_path)
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+ with open(output_path, "rb") as f:
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+ st.download_button(
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+ label="Download Audio",
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+ data=f,
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+ file_name="translated_audio.wav",
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+ mime="audio/wav"
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+ )
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+
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+ def main():
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+ """Main application workflow"""
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+ configure_page()
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+ st.title("🎧 High-Quality Audio Translation System")
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+ st.markdown("Upload English Audio → Get Chinese Speech Output")
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+
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+ # File uploader widget
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+ uploaded_file = st.file_uploader(
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+ "Select Audio File (MP3/WAV)",
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+ type=["mp3", "wav"],
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+ accept_multiple_files=False
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+ )
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+
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+ if uploaded_file:
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+ # Save uploaded file
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+ upload_path = os.path.join("temp/uploads", uploaded_file.name)
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+ with open(upload_path, "wb") as f:
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+ f.write(uploaded_file.getbuffer())
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+
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+ # Execute processing pipeline
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+ results = handle_file_processing(upload_path)
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+ if results:
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+ render_results(*results)
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+
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+ if __name__ == "__main__":
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+ main()
pyproject.toml ADDED
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+ [tool.poetry]
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+ name = "teaching-assistant"
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+ version = "0.1.0"
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+ description = "High-quality audio translation web application"
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+ authors = ["Your Name <[email protected]>"]
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+ license = "MIT"
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+ keywords = ["nlp", "translation", "speech-processing"]
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+
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+ [tool.poetry.dependencies]
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+ python = "^3.9"
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+
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+ # Core dependencies
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+ streamlit = "^1.31.1"
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+ pydub = "^0.25.1"
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+ python-dotenv = "^1.0.0"
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+ nltk = "^3.8.1" # Text segmentation
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+ librosa = "^0.10.1" # Advanced audio processing
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+ soundfile = "^0.12.1" # Audio file I/O
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+ ffmpeg-python = "^0.2.0" # FFmpeg integration
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+
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+ # Machine learning frameworks
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+ torch = { version = "^2.2.1", source = "pytorch" }
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+ transformers = { version = "^4.38.2", extras = ["audio"] }
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+
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+ # Text-to-speech engine
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+ TTS = "^0.21.0"
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+
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+ # Platform-specific dependencies
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+ torchaudio = { version = "^2.2.1", source = "pytorch", optional = true }
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+
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+ [tool.poetry.group.dev.dependencies]
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+ black = "^24.3.0"
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+ flake8 = "^6.1.0"
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+ mypy = "^1.8.0"
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+ pytest = "^8.0.2"
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+
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+ [[tool.poetry.source]]
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+ name = "pytorch"
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+ url = "https://download.pytorch.org/whl/cpu"
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+ priority = "primary"
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+
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+ [build-system]
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+ requires = ["poetry-core>=1.3.2"]
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+ build-backend = "poetry.core.masonry.api"
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+
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+ [tool.poetry.extras]
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+ gpu = ["torchaudio"]
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+
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+ [tool.poetry.scripts]
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+ start = "app:main"
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+
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+ [project.urls]
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+ Documentation = "https://github.com/yourusername/audio-translator/wiki"
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+ Issue-Tracker = "https://github.com/yourusername/audio-translator/issues"
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+
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+ # Configuration notes:
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+ # 1. Torch dependencies are sourced from PyTorch's official repository
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+ # 2. Transformers include audio processing extras
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+ # 3. GPU support can be enabled via: poetry install --extras "gpu"
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+ # 4. Platform-specific dependencies are handled through optional groups
utils/stt.py ADDED
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+ """
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+ Speech Recognition Module using Whisper Large-v3
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+ Handles audio preprocessing and transcription
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+ """
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+
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+ import torch
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+ from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor
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+ from pydub import AudioSegment
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+
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+ def transcribe_audio(audio_path):
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+ """
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+ Convert audio file to text using Whisper ASR model
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+ Args:
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+ audio_path: Path to input audio file
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+ Returns:
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+ Transcribed English text
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+ """
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+ # Configure hardware settings
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+ device = "cuda" if torch.cuda.is_available() else "cpu"
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+
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+ # Convert to proper audio format
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+ audio = AudioSegment.from_file(audio_path)
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+ processed_audio = audio.set_frame_rate(16000).set_channels(1)
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+ wav_path = audio_path.replace(".mp3", ".wav")
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+ processed_audio.export(wav_path, format="wav")
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+
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+ # Initialize ASR model
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+ model = AutoModelForSpeechSeq2Seq.from_pretrained(
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+ "openai/whisper-large-v3",
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+ torch_dtype=torch.float32,
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+ low_cpu_mem_usage=True,
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+ use_safetensors=True
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+ ).to(device)
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+
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+ processor = AutoProcessor.from_pretrained("openai/whisper-large-v3")
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+
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+ # Process audio input
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+ inputs = processor(
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+ wav_path,
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+ sampling_rate=16000,
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+ return_tensors="pt",
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+ truncation=True,
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+ chunk_length_s=30,
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+ stride_length_s=5
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+ ).to(device)
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+
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+ # Generate transcription
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+ with torch.no_grad():
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+ outputs = model.generate(**inputs, language="en", task="transcribe")
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+
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+ return processor.batch_decode(outputs, skip_special_tokens=True)[0]
utils/translation.py ADDED
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+ """
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+ Text Translation Module using NLLB-3.3B model
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+ Handles text segmentation and batch translation
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+ """
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+
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+ from transformers import AutoTokenizer, AutoModelForSeq2SeqLM
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+
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+ def translate_text(text):
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+ """
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+ Translate English text to Simplified Chinese
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+ Args:
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+ text: Input English text
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+ Returns:
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+ Translated Chinese text
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+ """
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+ # Initialize translation model
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+ tokenizer = AutoTokenizer.from_pretrained("facebook/nllb-200-3.3B")
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+ model = AutoModelForSeq2SeqLM.from_pretrained("facebook/nllb-200-3.3B")
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+
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+ # Split long text into manageable chunks
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+ max_chunk_length = 1000
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+ text_chunks = [
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+ text[i:i+max_chunk_length]
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+ for i in range(0, len(text), max_chunk_length)
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+ ]
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+
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+ translated_chunks = []
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+ for chunk in text_chunks:
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+ # Prepare model inputs
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+ inputs = tokenizer(
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+ chunk,
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+ return_tensors="pt",
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+ max_length=1024,
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+ truncation=True
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+ )
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+
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+ # Generate translation
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+ outputs = model.generate(
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+ **inputs,
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+ forced_bos_token_id=tokenizer.lang_code_to_id["zho_Hans"],
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+ max_new_tokens=1024
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+ )
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+ translated_chunks.append(tokenizer.decode(outputs[0], skip_special_tokens=True))
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+
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+ return "".join(translated_chunks)
utils/tts.py ADDED
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+ """
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+ Text-to-Speech Module using YourTTS
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+ Handles speech synthesis and output generation
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+ """
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+
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+ from TTS.api import TTS
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+ import os
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+ import time
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+
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+ def generate_speech(text):
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+ """
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+ Convert Chinese text to natural-sounding speech
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+ Args:
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+ text: Input Chinese text
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+ Returns:
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+ Path to generated audio file
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+ """
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+ # Initialize TTS engine
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+ tts = TTS(
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+ model_name="tts_models/multilingual/multi-dataset/your_tts",
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+ progress_bar=False,
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+ gpu=False
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+ )
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+
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+ # Create unique output filename
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+ output_path = os.path.join(
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+ "temp/outputs",
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+ f"output_{int(time.time())}.wav"
29
+ )
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+
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+ # Use reference voice if available
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+ ref_voice = (
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+ "assets/reference_voice.wav"
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+ if os.path.exists("assets/reference_voice.wav")
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+ else None
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+ )
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+
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+ # Generate speech output
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+ tts.tts_to_file(
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+ text=text,
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+ speaker_wav=ref_voice,
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+ language="zh-cn",
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+ file_path=output_path
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+ )
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+
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+ return output_path