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import gradio as gr
import asyncio
import websockets
import json
import logging
import time
from typing import Dict, Any, Optional
import threading
from queue import Queue
import base64
import numpy as np
import os
# Configure logging
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
# Environment-configurable HF Space URL (matching backend.py)
HF_SPACE_URL = os.getenv("HF_SPACE_URL", "https://androidguy-speaker-diarization.hf.space")
API_WS = f"wss://{HF_SPACE_URL}/ws_inference"
class TranscriptionWebSocketServer:
"""WebSocket server that receives audio from backend and returns transcription results"""
def __init__(self):
self.connected_clients = set()
self.is_running = False
self.websocket_server = None
self.conversation_history = []
self.processing_stats = {
"total_audio_chunks": 0,
"total_transcriptions": 0,
"last_audio_received": None,
"server_start_time": time.time(),
"backend_url": HF_SPACE_URL
}
async def handle_client_connection(self, websocket, path):
"""Handle incoming WebSocket connections from the backend"""
client_addr = websocket.remote_address
logger.info(f"Backend client connected from {client_addr}")
self.connected_clients.add(websocket)
try:
# Send initial connection acknowledgment
await websocket.send(json.dumps({
"type": "connection_ack",
"status": "connected",
"timestamp": time.time(),
"message": "HuggingFace transcription service ready"
}))
# Handle incoming messages/audio data
async for message in websocket:
try:
if isinstance(message, bytes):
# Handle binary audio data
await self.process_audio_data(message, websocket)
else:
# Handle text messages (JSON)
await self.handle_text_message(message, websocket)
except Exception as e:
logger.error(f"Error processing message: {e}")
await self.send_error(websocket, f"Processing error: {str(e)}")
except websockets.exceptions.ConnectionClosed:
logger.info("Backend client disconnected")
except Exception as e:
logger.error(f"Client connection error: {e}")
finally:
self.connected_clients.discard(websocket)
logger.info(f"Client removed. Active connections: {len(self.connected_clients)}")
async def process_audio_data(self, audio_data: bytes, websocket):
"""Process incoming audio data and return transcription results"""
try:
self.processing_stats["total_audio_chunks"] += 1
self.processing_stats["last_audio_received"] = time.time()
logger.debug(f"Received {len(audio_data)} bytes of audio data")
# Try to import and use your inference functions
try:
from inference import transcribe_audio, identify_speakers
# Process the audio for transcription
transcription_result = await transcribe_audio(audio_data)
if transcription_result:
# Process for speaker diarization if available
try:
speaker_info = await identify_speakers(audio_data)
transcription_result.update(speaker_info)
except Exception as e:
logger.warning(f"Speaker diarization failed: {e}")
transcription_result["speaker"] = "Unknown"
# Update conversation history
self.update_conversation_history(transcription_result)
# Send result back to backend
response = {
"type": "processing_result",
"timestamp": time.time(),
"data": transcription_result
}
await websocket.send(json.dumps(response))
self.processing_stats["total_transcriptions"] += 1
logger.info(f"Sent transcription result: {transcription_result.get('text', '')[:50]}...")
except ImportError:
# Fallback if inference module is not available
logger.warning("Inference module not found, using mock transcription")
# Try to use shared.py for processing if available
try:
from shared import RealtimeSpeakerDiarization
# Initialize if not already initialized
if not hasattr(self, 'diarization_system'):
self.diarization_system = RealtimeSpeakerDiarization()
await asyncio.to_thread(self.diarization_system.initialize_models)
await asyncio.to_thread(self.diarization_system.start_recording)
# Process the audio chunk
result = await asyncio.to_thread(self.diarization_system.process_audio_chunk, audio_data)
# Format result for response
if result and result["status"] != "error":
mock_result = {
"text": result.get("text", f"[Processing {len(audio_data)} bytes]"),
"speaker": f"Speaker_{result.get('speaker_id', 0) + 1}",
"confidence": result.get("similarity", 0.85),
"timestamp": time.time()
}
else:
# Fallback mock result
mock_result = {
"text": f"[Mock transcription - {len(audio_data)} bytes processed]",
"speaker": "Speaker_1",
"confidence": 0.85,
"timestamp": time.time()
}
# Update conversation history
self.update_conversation_history(mock_result)
response = {
"type": "processing_result",
"timestamp": time.time(),
"data": mock_result
}
await websocket.send(json.dumps(response))
self.processing_stats["total_transcriptions"] += 1
except Exception as e:
logger.warning(f"Failed to use shared module: {e}")
# Basic mock transcription as last resort
mock_result = {
"text": f"[Mock transcription - {len(audio_data)} bytes processed]",
"speaker": "Speaker_1",
"confidence": 0.85,
"timestamp": time.time()
}
self.update_conversation_history(mock_result)
response = {
"type": "processing_result",
"timestamp": time.time(),
"data": mock_result
}
await websocket.send(json.dumps(response))
except Exception as e:
logger.error(f"Audio processing error: {e}")
await self.send_error(websocket, f"Audio processing failed: {str(e)}")
async def handle_text_message(self, message: str, websocket):
"""Handle text-based messages from backend"""
try:
data = json.loads(message)
message_type = data.get("type", "unknown")
logger.info(f"Received message type: {message_type}")
if message_type == "ping":
# Respond to ping with pong
await websocket.send(json.dumps({
"type": "pong",
"timestamp": time.time()
}))
elif message_type == "config":
# Handle configuration updates
logger.info(f"Configuration update: {data}")
# Apply configuration settings if available
settings = data.get("settings", {})
if "max_speakers" in settings:
max_speakers = settings.get("max_speakers")
logger.info(f"Setting max_speakers to {max_speakers}")
if "threshold" in settings:
threshold = settings.get("threshold")
logger.info(f"Setting speaker change threshold to {threshold}")
# Send acknowledgment
await websocket.send(json.dumps({
"type": "config_ack",
"message": "Configuration received",
"timestamp": time.time()
}))
elif message_type == "status_request":
# Send status information
await websocket.send(json.dumps({
"type": "status_response",
"data": self.get_processing_stats(),
"timestamp": time.time()
}))
else:
logger.warning(f"Unknown message type: {message_type}")
except json.JSONDecodeError:
logger.error(f"Invalid JSON received: {message}")
await self.send_error(websocket, "Invalid JSON format")
async def send_error(self, websocket, error_message: str):
"""Send error message to client"""
try:
await websocket.send(json.dumps({
"type": "error",
"message": error_message,
"timestamp": time.time()
}))
except Exception as e:
logger.error(f"Failed to send error message: {e}")
def update_conversation_history(self, transcription_result: Dict[str, Any]):
"""Update conversation history with new transcription"""
history_entry = {
"timestamp": time.time(),
"text": transcription_result.get("text", ""),
"speaker": transcription_result.get("speaker", "Unknown"),
"confidence": transcription_result.get("confidence", 0.0)
}
self.conversation_history.append(history_entry)
# Keep only last 50 entries to prevent memory issues
if len(self.conversation_history) > 50:
self.conversation_history = self.conversation_history[-50:]
def get_processing_stats(self):
"""Get processing statistics"""
return {
"connected_clients": len(self.connected_clients),
"total_audio_chunks": self.processing_stats["total_audio_chunks"],
"total_transcriptions": self.processing_stats["total_transcriptions"],
"last_audio_received": self.processing_stats["last_audio_received"],
"server_uptime": time.time() - self.processing_stats["server_start_time"],
"conversation_entries": len(self.conversation_history),
"backend_url": self.processing_stats.get("backend_url", HF_SPACE_URL)
}
async def start_server(self, host="0.0.0.0", port=7860):
"""Start the WebSocket server"""
try:
# Start WebSocket server on /ws_inference endpoint
self.websocket_server = await websockets.serve(
self.handle_client_connection,
host,
port,
subprotocols=[],
path="/ws_inference"
)
self.is_running = True
logger.info(f"WebSocket server started on ws://{host}:{port}/ws_inference")
# Keep the server running
await self.websocket_server.wait_closed()
except Exception as e:
logger.error(f"Failed to start WebSocket server: {e}")
self.is_running = False
# Initialize the WebSocket server
ws_server = TranscriptionWebSocketServer()
def create_gradio_interface():
"""Create Gradio interface for monitoring and testing"""
def get_server_status():
"""Get current server status"""
stats = ws_server.get_processing_stats()
status_text = f"""
### Server Status
- **WebSocket Server**: {'🟒 Running' if ws_server.is_running else 'πŸ”΄ Stopped'}
- **Connected Clients**: {stats['connected_clients']}
- **Server Uptime**: {stats['server_uptime']:.1f} seconds
### Processing Statistics
- **Audio Chunks Processed**: {stats['total_audio_chunks']}
- **Transcriptions Generated**: {stats['total_transcriptions']}
- **Last Audio Received**: {time.ctime(stats['last_audio_received']) if stats['last_audio_received'] else 'Never'}
### Conversation
- **History Entries**: {stats['conversation_entries']}
"""
return status_text
def get_recent_transcriptions():
"""Get recent transcription results"""
if not ws_server.conversation_history:
return "No transcriptions yet. Waiting for audio data from backend..."
recent_entries = ws_server.conversation_history[-10:] # Last 10 entries
formatted_text = "### Recent Transcriptions\n\n"
for entry in recent_entries:
timestamp = time.strftime("%H:%M:%S", time.localtime(entry['timestamp']))
speaker = entry['speaker']
text = entry['text']
confidence = entry['confidence']
# Extract speaker number for color matching with shared.py
speaker_num = 0
if speaker.startswith("Speaker_"):
try:
speaker_num = int(speaker.split("_")[1]) - 1
except (ValueError, IndexError):
speaker_num = 0
# Use colors from shared.py if possible
try:
from shared import SPEAKER_COLORS
color = SPEAKER_COLORS[speaker_num % len(SPEAKER_COLORS)]
except (ImportError, IndexError):
# Fallback colors
colors = ["#FF6B6B", "#4ECDC4", "#45B7D1", "#96CEB4", "#FFEAA7", "#DDA0DD", "#98D8C8", "#F7DC6F"]
color = colors[speaker_num % len(colors)]
formatted_text += f"<span style='color:{color};font-weight:bold;'>[{timestamp}] {speaker}</span> (confidence: {confidence:.2f})\n"
formatted_text += f"{text}\n\n"
return formatted_text
def clear_conversation_history():
"""Clear conversation history"""
ws_server.conversation_history.clear()
return "Conversation history cleared!"
# Create Gradio interface
with gr.Blocks(
title="Real-time Audio Transcription Service",
theme=gr.themes.Soft()
) as demo:
gr.Markdown("# 🎀 Real-time Audio Transcription Service")
gr.Markdown("This HuggingFace Space receives audio from your backend and returns transcription results with speaker diarization.")
with gr.Tab("πŸ“Š Server Status"):
status_display = gr.Markdown(get_server_status())
with gr.Row():
refresh_status_btn = gr.Button("πŸ”„ Refresh Status", variant="primary")
refresh_status_btn.click(
fn=get_server_status,
outputs=status_display,
every=None
)
with gr.Tab("πŸ“ Live Transcription"):
transcription_display = gr.Markdown(get_recent_transcriptions())
with gr.Row():
refresh_transcription_btn = gr.Button("πŸ”„ Refresh Transcriptions", variant="primary")
clear_history_btn = gr.Button("πŸ—‘οΈ Clear History", variant="secondary")
refresh_transcription_btn.click(
fn=get_recent_transcriptions,
outputs=transcription_display
)
clear_history_btn.click(
fn=clear_conversation_history,
outputs=gr.Markdown()
)
with gr.Tab("πŸ”§ Connection Info"):
gr.Markdown(f"""
### WebSocket Connection Details
**WebSocket Endpoint**: `wss://{HF_SPACE_URL}/ws_inference`
### Backend Connection
Your backend should connect to this WebSocket endpoint and:
1. **Send Audio Data**: Stream raw audio bytes to this endpoint
2. **Receive Results**: Get JSON responses with transcription results
### Expected Message Flow
**Backend β†’ HuggingFace**:
- Raw audio bytes (binary data)
- Configuration messages (JSON)
**HuggingFace β†’ Backend**:
```json
{{
"type": "processing_result",
"timestamp": 1234567890.123,
"data": {{
"text": "transcribed text here",
"speaker": "Speaker_1",
"confidence": 0.95
}}
}}
```
### Test Connection
Your backend is configured to connect to: `{ws_server.processing_stats.get('backend_url', HF_SPACE_URL)}`
""")
with gr.Tab("πŸš€ API Documentation"):
gr.Markdown("""
### WebSocket API Reference
#### Endpoint
- **URL**: `/ws_inference`
- **Protocol**: WebSocket
- **Accepts**: Binary audio data + JSON messages
#### Message Types
##### 1. Audio Processing
- **Input**: Raw audio bytes (binary)
- **Output**: Processing result (JSON)
##### 2. Configuration
- **Input**:
```json
{
"type": "config",
"settings": {
"language": "en",
"enable_diarization": true,
"max_speakers": 4,
"threshold": 0.65
}
}
```
##### 3. Status Check
- **Input**: `{"type": "status_request"}`
- **Output**: Server statistics
##### 4. Ping/Pong
- **Input**: `{"type": "ping"}`
- **Output**: `{"type": "pong", "timestamp": 1234567890}`
#### Error Handling
All errors are returned as:
```json
{
"type": "error",
"message": "Error description",
"timestamp": 1234567890.123
}
```
""")
return demo
def run_websocket_server():
"""Run WebSocket server in background thread"""
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
try:
logger.info("Starting WebSocket server thread...")
loop.run_until_complete(ws_server.start_server())
except Exception as e:
logger.error(f"WebSocket server error: {e}")
finally:
loop.close()
# Mount UI to inference.py
def mount_ui(app):
"""Mount Gradio interface to FastAPI app"""
try:
demo = create_gradio_interface()
# Mount without starting server (FastAPI will handle it)
demo.mount_to_app(app)
logger.info("Gradio UI mounted to FastAPI app")
return True
except Exception as e:
logger.error(f"Error mounting UI: {e}")
return False
# Start WebSocket server in background
logger.info("Initializing WebSocket server...")
websocket_thread = threading.Thread(target=run_websocket_server, daemon=True)
websocket_thread.start()
# Give server time to start
time.sleep(2)
# Create and launch Gradio interface
if __name__ == "__main__":
demo = create_gradio_interface()
demo.launch(
server_name="0.0.0.0",
server_port=7860,
share=True,
show_error=True
)