Spaces:
Runtime error
Runtime error
first commit
Browse files- .gitignore +55 -0
- client.py +280 -0
- main.py +220 -0
- requirements.txt +13 -0
.gitignore
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# Byte-compiled / optimized / DLL files
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__pycache__/
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*.py[cod]
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*$py.class
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# Model files (these will be downloaded at runtime)
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models/*.ckpt
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models/*.yaml
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models/*.pt
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models/*.bin
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# Audio files (generated content)
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audio_files/
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*.wav
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# Distribution / packaging
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dist/
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build/
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*.egg-info/
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# Virtual environments
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venv/
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env/
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ENV/
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.env
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.venv
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# IDE specific files
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.idea/
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.vscode/
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*.swp
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*.swo
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# Jupyter Notebook
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.ipynb_checkpoints
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# OS specific files
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.DS_Store
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Thumbs.db
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# Logs
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*.log
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logs/
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# Local configuration
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config.local.py
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.env.local
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# Temporary files
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tmp/
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temp/
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# Cache directories
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.cache/
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.pytest_cache/
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client.py
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import streamlit as st
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import requests
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import base64
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from io import BytesIO
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import pandas as pd
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# Set page config
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st.set_page_config(
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page_title="Nigerian Text-to-Speech",
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page_icon="🎙️",
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layout="wide"
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)
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# Define the available voices and languages
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AVAILABLE_VOICES = {
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"Female": ["zainab", "idera", "regina", "chinenye", "joke", "remi"],
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"Male": ["jude", "tayo", "umar", "osagie", "onye", "emma"]
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}
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AVAILABLE_LANGUAGES = ["english", "yoruba", "igbo", "hausa"]
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# IMPORTANT: Replace this with the ngrok URL shown in your Colab notebook
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# Example: API_BASE_URL = "https://a1b2-34-56-78-90.ngrok.io"
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API_BASE_URL = st.text_input(
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"Enter the ngrok URL from Colab (e.g., https://a1b2-34-56-78-90.ngrok.io)",
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value="",
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key="api_url"
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)
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# Derive the TTS endpoint from the base URL
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if API_BASE_URL:
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API_TTS_ENDPOINT = f"{API_BASE_URL}/tts"
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# Test connection to backend
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try:
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health_check = requests.get(f"{API_BASE_URL}")
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if health_check.status_code == 200:
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st.success(f"✅ Connected to backend API successfully!")
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else:
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st.warning(f"⚠️ Backend API returned status code {health_check.status_code}")
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except Exception as e:
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st.error(f"❌ Cannot connect to backend API: {str(e)}")
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else:
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st.warning("⚠️ Please enter the ngrok URL from your Colab notebook to continue")
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# App title and description
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st.title("Nigerian Text-to-Speech")
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st.markdown("""
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Convert text to speech with authentic Nigerian accents. This app uses YarnGPT, a text-to-speech model
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that generates natural Nigerian-accented speech in English, Yoruba, Igbo, and Hausa.
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""")
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# Create tabs for different functions
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tab1, tab2, tab3 = st.tabs(["Basic TTS", "Batch Processing", "About"])
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# Tab 1: Basic TTS
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with tab1:
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col1, col2 = st.columns([3, 1])
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with col1:
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# Text input
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text_input = st.text_area(
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"Enter text to convert to speech",
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"Welcome to Nigeria, the giant of Africa. Our diverse cultures and languages make us unique.",
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height=150
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)
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# Generate button
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generate_button = st.button("Generate Audio", type="primary", disabled=not API_BASE_URL)
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with col2:
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# Options
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language = st.selectbox("Language", AVAILABLE_LANGUAGES)
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gender = st.radio("Gender", ["Female", "Male"])
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voice = st.selectbox("Voice", AVAILABLE_VOICES[gender])
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st.info(f"Selected voice: **{voice}** ({gender.lower()})")
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# Generate audio when button is clicked
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if generate_button and text_input and API_BASE_URL:
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with st.spinner("Generating audio... (This may take a minute as the audio is processed through Colab)"):
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try:
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# Call the API with timeout increased
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response = requests.post(
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API_TTS_ENDPOINT,
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json={"text": text_input, "language": language, "voice": voice},
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timeout=100000 # Increase timeout to 2 minutes
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)
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if response.status_code == 200:
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# Get response data
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audio_data = response.json()
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# Save info in session state
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st.session_state.last_text = text_input
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st.session_state.last_voice = voice
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st.session_state.last_language = language
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# Display success and audio player
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st.success("Audio generated successfully!")
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st.markdown(f"Voice: **{voice}** | Language: **{language}**")
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# Handle base64-encoded audio
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if "audio_base64" in audio_data:
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audio_bytes = base64.b64decode(audio_data["audio_base64"])
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audio_stream = BytesIO(audio_bytes)
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# Play audio directly from the stream
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st.audio(audio_stream, format="audio/wav")
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else:
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# Fall back to URL method (legacy support)
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audio_url = f"{API_BASE_URL}{audio_data['audio_url']}"
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st.warning("Using legacy URL-based audio (may not work)")
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st.code(audio_url, language="text")
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st.audio(audio_url, format="audio/wav")
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else:
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st.error(f"Error: {response.status_code} - {response.text}")
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except Exception as e:
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st.error(f"Error generating audio: {str(e)}")
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st.info(f"Make sure the backend API is running and accessible at {API_BASE_URL}")
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# Tab 2: Batch Processing
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with tab2:
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st.header("Batch Text-to-Speech Conversion")
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st.markdown("""
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Process multiple text entries at once. Upload a CSV file with the following columns:
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- `text`: The text to convert to speech
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- `language` (optional): Language for the text (english, yoruba, igbo, hausa)
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- `voice` (optional): Voice name to use
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""")
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# File uploader
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uploaded_file = st.file_uploader("Upload CSV file", type="csv")
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if uploaded_file and API_BASE_URL:
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# Process the file
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try:
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df = pd.read_csv(uploaded_file)
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if "text" not in df.columns:
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st.error("CSV file must contain a 'text' column")
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else:
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st.dataframe(df.head())
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# Default values
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default_language = st.selectbox("Default language", AVAILABLE_LANGUAGES)
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default_voice = st.selectbox("Default voice", AVAILABLE_VOICES["Female"] + AVAILABLE_VOICES["Male"])
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if st.button("Process Batch", disabled=not API_BASE_URL):
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# Create a container for audio files
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audio_container = st.container()
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progress_bar = st.progress(0)
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status_text = st.empty()
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# Process each row
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results = []
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audio_files = [] # Store audio data for playback
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for i, row in enumerate(df.itertuples()):
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# Update progress
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progress = int((i + 1) / len(df) * 100)
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progress_bar.progress(progress)
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status_text.text(f"Processing item {i+1} of {len(df)}...")
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+
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# Get text and parameters
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text = row.text
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lang = getattr(row, 'language', default_language) if hasattr(row, 'language') else default_language
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voice_name = getattr(row, 'voice', default_voice) if hasattr(row, 'voice') else default_voice
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169 |
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try:
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# Make API call with increased timeout
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response = requests.post(
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API_TTS_ENDPOINT,
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json={"text": text, "language": lang, "voice": voice_name},
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timeout=120 # Increase timeout to 2 minutes
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)
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177 |
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178 |
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if response.status_code == 200:
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179 |
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audio_data = response.json()
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180 |
+
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# Handle base64-encoded audio
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182 |
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if "audio_base64" in audio_data:
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audio_bytes = base64.b64decode(audio_data["audio_base64"])
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184 |
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audio_files.append({
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"index": i,
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"bytes": audio_bytes,
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187 |
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"text": text,
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188 |
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"voice": voice_name,
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189 |
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"language": lang
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})
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191 |
+
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status = "Success"
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else:
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# Fall back to URL method (legacy support)
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195 |
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audio_url = f"{API_BASE_URL}{audio_data['audio_url']}"
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status = "Success (URL mode)"
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197 |
+
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198 |
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# Add to results
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199 |
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results.append({
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200 |
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"text": text[:50] + "..." if len(text) > 50 else text,
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201 |
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"language": lang,
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202 |
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"voice": voice_name,
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203 |
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"status": status
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})
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205 |
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else:
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results.append({
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"text": text[:50] + "..." if len(text) > 50 else text,
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"language": lang,
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"voice": voice_name,
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"status": f"Error: {response.status_code}"
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})
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except Exception as e:
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results.append({
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"text": text[:50] + "..." if len(text) > 50 else text,
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"language": lang,
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"voice": voice_name,
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"status": f"Error: {str(e)}"
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})
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# Show results
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st.success("Batch processing completed!")
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results_df = pd.DataFrame(results)
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st.dataframe(results_df)
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# Display audio players for successful generations
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with audio_container:
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st.subheader("Generated Audio Files")
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228 |
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for audio_item in audio_files:
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st.markdown(f"**{audio_item['index']+1}. {audio_item['text'][:50]}...** ({audio_item['voice']}, {audio_item['language']})")
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audio_stream = BytesIO(audio_item["bytes"])
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231 |
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st.audio(audio_stream, format="audio/wav")
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232 |
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st.markdown("---")
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233 |
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except Exception as e:
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st.error(f"Error processing file: {str(e)}")
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236 |
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elif not API_BASE_URL:
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237 |
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st.warning("Please enter the ngrok URL first to enable batch processing")
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239 |
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# Tab 3: About
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with tab3:
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st.header("About YarnGPT")
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col1, col2 = st.columns([1, 1])
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with col1:
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st.markdown("""
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### Features
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248 |
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- 🗣️ 12 preset voices (6 male, 6 female)
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- 🎯 Trained on 2000+ hours of Nigerian audio
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250 |
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- 🔊 24kHz high-quality audio output
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251 |
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- 📝 Support for long-form text
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252 |
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253 |
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### Model Details
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254 |
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- Base: HuggingFaceTB/SmolLM2-360M
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255 |
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- Training: 5 epochs on A100 GPU
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256 |
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- Data: Nigerian movies, podcasts, and open-source audio
|
257 |
+
""")
|
258 |
+
|
259 |
+
with col2:
|
260 |
+
st.markdown("""
|
261 |
+
### Available Voices
|
262 |
+
- **Female**: zainab, idera, regina, chinenye, joke, remi
|
263 |
+
- **Male**: jude, tayo, umar, osagie, onye, emma
|
264 |
+
|
265 |
+
### Limitations
|
266 |
+
- English to Nigerian-accented English primarily
|
267 |
+
- May not capture all Nigerian accent variations
|
268 |
+
- Training data includes auto-generated content
|
269 |
+
""")
|
270 |
+
|
271 |
+
st.markdown("""
|
272 |
+
### Credits
|
273 |
+
- YarnGPT was created by Saheed Abdulrahman, a Unilag student
|
274 |
+
- Model is available as open source on [GitHub](https://github.com/saheedniyi02/yarngpt)
|
275 |
+
- Web demo: [https://yarngpt.co/](https://yarngpt.co/)
|
276 |
+
""")
|
277 |
+
|
278 |
+
# Footer
|
279 |
+
st.markdown("---")
|
280 |
+
st.markdown("Developed for a Nigerian News App Podcaster API | Powered by YarnGPT")
|
main.py
ADDED
@@ -0,0 +1,220 @@
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
from fastapi import FastAPI, HTTPException, BackgroundTasks
|
2 |
+
from fastapi.responses import StreamingResponse
|
3 |
+
from fastapi.middleware.cors import CORSMiddleware
|
4 |
+
from pydantic import BaseModel
|
5 |
+
import os
|
6 |
+
import uuid
|
7 |
+
import torch
|
8 |
+
import torchaudio
|
9 |
+
import base64
|
10 |
+
from io import BytesIO
|
11 |
+
from transformers import AutoModelForCausalLM
|
12 |
+
import sys
|
13 |
+
import subprocess
|
14 |
+
from datetime import datetime, timedelta
|
15 |
+
|
16 |
+
app = FastAPI(title="Nigerian TTS API")
|
17 |
+
|
18 |
+
# Add CORS middleware
|
19 |
+
app.add_middleware(
|
20 |
+
CORSMiddleware,
|
21 |
+
allow_origins=["*"], # In production, set this to your Next.js domain
|
22 |
+
allow_credentials=True,
|
23 |
+
allow_methods=["*"],
|
24 |
+
allow_headers=["*"],
|
25 |
+
)
|
26 |
+
|
27 |
+
# Initialize necessary directories
|
28 |
+
os.makedirs("audio_files", exist_ok=True)
|
29 |
+
os.makedirs("models", exist_ok=True)
|
30 |
+
|
31 |
+
# Check if YarnGPT is installed, if not install it
|
32 |
+
try:
|
33 |
+
import yarngpt
|
34 |
+
from yarngpt.audiotokenizer import AudioTokenizerV2
|
35 |
+
except ImportError:
|
36 |
+
print("Installing YarnGPT and dependencies...")
|
37 |
+
subprocess.check_call([sys.executable, "-m", "pip", "install", "git+https://github.com/saheedniyi02/yarngpt.git"])
|
38 |
+
subprocess.check_call([sys.executable, "-m", "pip", "install", "outetts", "uroman", "transformers", "torchaudio"])
|
39 |
+
from yarngpt.audiotokenizer import AudioTokenizerV2
|
40 |
+
|
41 |
+
# Model configuration
|
42 |
+
tokenizer_path = "saheedniyi/YarnGPT2"
|
43 |
+
|
44 |
+
# Check if model files exist, if not download them
|
45 |
+
wav_tokenizer_config_path = "./models/wavtokenizer_mediumdata_frame75_3s_nq1_code4096_dim512_kmeans200_attn.yaml"
|
46 |
+
wav_tokenizer_model_path = "./models/wavtokenizer_large_speech_320_24k.ckpt"
|
47 |
+
|
48 |
+
if not os.path.exists(wav_tokenizer_config_path):
|
49 |
+
print("Downloading model config file...")
|
50 |
+
subprocess.check_call([
|
51 |
+
"wget", "-O", wav_tokenizer_config_path,
|
52 |
+
"https://huggingface.co/novateur/WavTokenizer-medium-speech-75token/resolve/main/wavtokenizer_mediumdata_frame75_3s_nq1_code4096_dim512_kmeans200_attn.yaml"
|
53 |
+
])
|
54 |
+
|
55 |
+
if not os.path.exists(wav_tokenizer_model_path):
|
56 |
+
print("Downloading model checkpoint file...")
|
57 |
+
subprocess.check_call([
|
58 |
+
"wget", "-O", wav_tokenizer_model_path,
|
59 |
+
"https://drive.google.com/uc?id=1-ASeEkrn4HY49yZWHTASgfGFNXdVnLTt&export=download"
|
60 |
+
])
|
61 |
+
|
62 |
+
print("Loading YarnGPT model and tokenizer...")
|
63 |
+
audio_tokenizer = AudioTokenizerV2(
|
64 |
+
tokenizer_path, wav_tokenizer_model_path, wav_tokenizer_config_path
|
65 |
+
)
|
66 |
+
model = AutoModelForCausalLM.from_pretrained(tokenizer_path, torch_dtype="auto").to(audio_tokenizer.device)
|
67 |
+
print("Model loaded successfully!")
|
68 |
+
|
69 |
+
# Available voices and languages
|
70 |
+
AVAILABLE_VOICES = {
|
71 |
+
"female": ["zainab", "idera", "regina", "chinenye", "joke", "remi"],
|
72 |
+
"male": ["jude", "tayo", "umar", "osagie", "onye", "emma"]
|
73 |
+
}
|
74 |
+
AVAILABLE_LANGUAGES = ["english", "yoruba", "igbo", "hausa"]
|
75 |
+
|
76 |
+
# Input validation model
|
77 |
+
class TTSRequest(BaseModel):
|
78 |
+
text: str
|
79 |
+
language: str = "english"
|
80 |
+
voice: str = "idera"
|
81 |
+
|
82 |
+
# Output model with base64-encoded audio
|
83 |
+
class TTSResponse(BaseModel):
|
84 |
+
audio_base64: str # Base64-encoded audio data
|
85 |
+
audio_url: str # Keep for backward compatibility
|
86 |
+
text: str
|
87 |
+
voice: str
|
88 |
+
language: str
|
89 |
+
|
90 |
+
@app.get("/")
|
91 |
+
async def root():
|
92 |
+
"""API health check and info"""
|
93 |
+
return {
|
94 |
+
"status": "ok",
|
95 |
+
"message": "Nigerian TTS API is running",
|
96 |
+
"available_languages": AVAILABLE_LANGUAGES,
|
97 |
+
"available_voices": AVAILABLE_VOICES
|
98 |
+
}
|
99 |
+
|
100 |
+
|
101 |
+
@app.post("/tts", response_model=TTSResponse)
|
102 |
+
async def text_to_speech(request: TTSRequest, background_tasks: BackgroundTasks):
|
103 |
+
"""Convert text to Nigerian-accented speech"""
|
104 |
+
|
105 |
+
# Validate inputs
|
106 |
+
if request.language not in AVAILABLE_LANGUAGES:
|
107 |
+
raise HTTPException(status_code=400, detail=f"Language must be one of {AVAILABLE_LANGUAGES}")
|
108 |
+
|
109 |
+
all_voices = AVAILABLE_VOICES["female"] + AVAILABLE_VOICES["male"]
|
110 |
+
if request.voice not in all_voices:
|
111 |
+
raise HTTPException(status_code=400, detail=f"Voice must be one of {all_voices}")
|
112 |
+
|
113 |
+
# Generate unique filename
|
114 |
+
audio_id = str(uuid.uuid4())
|
115 |
+
output_path = f"audio_files/{audio_id}.wav"
|
116 |
+
|
117 |
+
try:
|
118 |
+
# Create prompt and generate audio
|
119 |
+
prompt = audio_tokenizer.create_prompt(request.text, lang=request.language, speaker_name=request.voice)
|
120 |
+
input_ids = audio_tokenizer.tokenize_prompt(prompt)
|
121 |
+
|
122 |
+
output = model.generate(
|
123 |
+
input_ids=input_ids,
|
124 |
+
temperature=0.1,
|
125 |
+
repetition_penalty=1.1,
|
126 |
+
max_length=4000,
|
127 |
+
)
|
128 |
+
|
129 |
+
codes = audio_tokenizer.get_codes(output)
|
130 |
+
audio = audio_tokenizer.get_audio(codes)
|
131 |
+
|
132 |
+
# Save audio file
|
133 |
+
torchaudio.save(output_path, audio, sample_rate=24000)
|
134 |
+
|
135 |
+
# Read the file and encode as base64
|
136 |
+
with open(output_path, "rb") as audio_file:
|
137 |
+
audio_bytes = audio_file.read()
|
138 |
+
audio_base64 = base64.b64encode(audio_bytes).decode('utf-8')
|
139 |
+
|
140 |
+
# Clean up old files after a while
|
141 |
+
background_tasks.add_task(cleanup_old_files)
|
142 |
+
|
143 |
+
return TTSResponse(
|
144 |
+
audio_base64=audio_base64,
|
145 |
+
audio_url=f"/audio/{audio_id}.wav", # Keep for compatibility
|
146 |
+
text=request.text,
|
147 |
+
voice=request.voice,
|
148 |
+
language=request.language
|
149 |
+
)
|
150 |
+
|
151 |
+
except Exception as e:
|
152 |
+
raise HTTPException(status_code=500, detail=f"Error generating audio: {str(e)}")
|
153 |
+
|
154 |
+
# File serving endpoint for direct audio access
|
155 |
+
@app.get("/audio/{filename}")
|
156 |
+
async def get_audio(filename: str):
|
157 |
+
file_path = f"audio_files/{filename}"
|
158 |
+
if not os.path.exists(file_path):
|
159 |
+
raise HTTPException(status_code=404, detail="Audio file not found")
|
160 |
+
|
161 |
+
def iterfile():
|
162 |
+
with open(file_path, "rb") as audio_file:
|
163 |
+
yield from audio_file
|
164 |
+
|
165 |
+
return StreamingResponse(iterfile(), media_type="audio/wav")
|
166 |
+
|
167 |
+
# Endpoint to stream audio directly from base64 (useful for debugging)
|
168 |
+
@app.post("/stream-audio")
|
169 |
+
async def stream_audio(request: TTSRequest):
|
170 |
+
"""Stream audio directly without saving to disk"""
|
171 |
+
try:
|
172 |
+
# Create prompt and generate audio
|
173 |
+
prompt = audio_tokenizer.create_prompt(request.text, lang=request.language, speaker_name=request.voice)
|
174 |
+
input_ids = audio_tokenizer.tokenize_prompt(prompt)
|
175 |
+
|
176 |
+
output = model.generate(
|
177 |
+
input_ids=input_ids,
|
178 |
+
temperature=0.1,
|
179 |
+
repetition_penalty=1.1,
|
180 |
+
max_length=4000,
|
181 |
+
)
|
182 |
+
|
183 |
+
codes = audio_tokenizer.get_codes(output)
|
184 |
+
audio = audio_tokenizer.get_audio(codes)
|
185 |
+
|
186 |
+
# Create BytesIO object
|
187 |
+
buffer = BytesIO()
|
188 |
+
torchaudio.save(buffer, audio, sample_rate=24000, format="wav")
|
189 |
+
buffer.seek(0)
|
190 |
+
|
191 |
+
return StreamingResponse(buffer, media_type="audio/wav")
|
192 |
+
except Exception as e:
|
193 |
+
raise HTTPException(status_code=500, detail=f"Error generating audio: {str(e)}")
|
194 |
+
|
195 |
+
# Cleanup function to remove old files
|
196 |
+
def cleanup_old_files():
|
197 |
+
"""Delete audio files older than 6 hours to manage disk space"""
|
198 |
+
try:
|
199 |
+
now = datetime.now()
|
200 |
+
audio_dir = "audio_files"
|
201 |
+
|
202 |
+
for filename in os.listdir(audio_dir):
|
203 |
+
if not filename.endswith(".wav"):
|
204 |
+
continue
|
205 |
+
|
206 |
+
file_path = os.path.join(audio_dir, filename)
|
207 |
+
file_mod_time = datetime.fromtimestamp(os.path.getmtime(file_path))
|
208 |
+
|
209 |
+
# Delete files older than 6 hours
|
210 |
+
if now - file_mod_time > timedelta(hours=6):
|
211 |
+
os.remove(file_path)
|
212 |
+
print(f"Deleted old audio file: {filename}")
|
213 |
+
except Exception as e:
|
214 |
+
print(f"Error cleaning up old files: {e}")
|
215 |
+
|
216 |
+
# For running locally with uvicorn
|
217 |
+
if __name__ == "__main__":
|
218 |
+
import uvicorn
|
219 |
+
port = int(os.environ.get("PORT", 8000))
|
220 |
+
uvicorn.run(app, host="0.0.0.0", port=port)
|
requirements.txt
ADDED
@@ -0,0 +1,13 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
fastapi==0.104.1
|
2 |
+
uvicorn==0.24.0
|
3 |
+
torch==2.1.0
|
4 |
+
torchaudio==2.1.0
|
5 |
+
transformers==4.35.0
|
6 |
+
pydantic==2.4.2
|
7 |
+
python-multipart==0.0.6
|
8 |
+
wget
|
9 |
+
gdown
|
10 |
+
numpy>=1.20.0
|
11 |
+
requests>=2.27.1
|
12 |
+
outetts
|
13 |
+
uroman
|