app.py

import gradio as gr
import numpy as np
import queue
import torch
import time
import threading
import os
import urllib.request
import torchaudio
from scipy.spatial.distance import cosine
from RealtimeSTT import AudioToTextRecorder
from fastapi import FastAPI
from fastrtc import Stream, AsyncStreamHandler, ReplyOnPause, get_cloudflare_turn_credentials_async, get_cloudflare_turn_credentials
import json
import io
import wave
import asyncio
import uvicorn

# Simplified configuration parameters
SILENCE_THRESHS = [0, 0.4]
FINAL_TRANSCRIPTION_MODEL = "distil-large-v3"
FINAL_BEAM_SIZE = 5
REALTIME_TRANSCRIPTION_MODEL = "distil-small.en"
REALTIME_BEAM_SIZE = 5
TRANSCRIPTION_LANGUAGE = "en"
SILERO_SENSITIVITY = 0.4
WEBRTC_SENSITIVITY = 3
MIN_LENGTH_OF_RECORDING = 0.7
PRE_RECORDING_BUFFER_DURATION = 0.35

# Speaker change detection parameters
DEFAULT_CHANGE_THRESHOLD = 0.7
EMBEDDING_HISTORY_SIZE = 5
MIN_SEGMENT_DURATION = 1.0
DEFAULT_MAX_SPEAKERS = 4
ABSOLUTE_MAX_SPEAKERS = 10

# Global variables
FAST_SENTENCE_END = True
SAMPLE_RATE = 16000
BUFFER_SIZE = 512
CHANNELS = 1

# Speaker colors
SPEAKER_COLORS = [
    "#FFFF00",  # Yellow
    "#FF0000",  # Red
    "#00FF00",  # Green
    "#00FFFF",  # Cyan
    "#FF00FF",  # Magenta
    "#0000FF",  # Blue
    "#FF8000",  # Orange
    "#00FF80",  # Spring Green
    "#8000FF",  # Purple
    "#FFFFFF",  # White
]

SPEAKER_COLOR_NAMES = [
    "Yellow", "Red", "Green", "Cyan", "Magenta", 
    "Blue", "Orange", "Spring Green", "Purple", "White"
]


class SpeechBrainEncoder:
    """ECAPA-TDNN encoder from SpeechBrain for speaker embeddings"""
    def __init__(self, device="cpu"):
        self.device = device
        self.model = None
        self.embedding_dim = 192
        self.model_loaded = False
        self.cache_dir = os.path.join(os.path.expanduser("~"), ".cache", "speechbrain")
        os.makedirs(self.cache_dir, exist_ok=True)
    
    def _download_model(self):
        """Download pre-trained SpeechBrain ECAPA-TDNN model if not present"""
        model_url = "https://huggingface.co/speechbrain/spkrec-ecapa-voxceleb/resolve/main/embedding_model.ckpt"
        model_path = os.path.join(self.cache_dir, "embedding_model.ckpt")
        
        if not os.path.exists(model_path):
            print(f"Downloading ECAPA-TDNN model to {model_path}...")
            urllib.request.urlretrieve(model_url, model_path)
        
        return model_path
    
    def load_model(self):
        """Load the ECAPA-TDNN model"""
        try:
            from speechbrain.pretrained import EncoderClassifier
            
            model_path = self._download_model()
            
            self.model = EncoderClassifier.from_hparams(
                source="speechbrain/spkrec-ecapa-voxceleb",
                savedir=self.cache_dir,
                run_opts={"device": self.device}
            )
            
            self.model_loaded = True
            return True
        except Exception as e:
            print(f"Error loading ECAPA-TDNN model: {e}")
            return False
    
    def embed_utterance(self, audio, sr=16000):
        """Extract speaker embedding from audio"""
        if not self.model_loaded:
            raise ValueError("Model not loaded. Call load_model() first.")
        
        try:
            if isinstance(audio, np.ndarray):
                waveform = torch.tensor(audio, dtype=torch.float32).unsqueeze(0)
            else:
                waveform = audio.unsqueeze(0)
            
            if sr != 16000:
                waveform = torchaudio.functional.resample(waveform, orig_freq=sr, new_freq=16000)
            
            with torch.no_grad():
                embedding = self.model.encode_batch(waveform)
                
            return embedding.squeeze().cpu().numpy()
        except Exception as e:
            print(f"Error extracting embedding: {e}")
            return np.zeros(self.embedding_dim)


class AudioProcessor:
    """Processes audio data to extract speaker embeddings"""
    def __init__(self, encoder):
        self.encoder = encoder
    
    def extract_embedding(self, audio_int16):
        try:
            float_audio = audio_int16.astype(np.float32) / 32768.0
            
            if np.abs(float_audio).max() > 1.0:
                float_audio = float_audio / np.abs(float_audio).max()
            
            embedding = self.encoder.embed_utterance(float_audio)
            
            return embedding
        except Exception as e:
            print(f"Embedding extraction error: {e}")
            return np.zeros(self.encoder.embedding_dim)


class SpeakerChangeDetector:
    """Speaker change detector that supports a configurable number of speakers"""
    def __init__(self, embedding_dim=192, change_threshold=DEFAULT_CHANGE_THRESHOLD, max_speakers=DEFAULT_MAX_SPEAKERS):
        self.embedding_dim = embedding_dim
        self.change_threshold = change_threshold
        self.max_speakers = min(max_speakers, ABSOLUTE_MAX_SPEAKERS)
        self.current_speaker = 0
        self.previous_embeddings = []
        self.last_change_time = time.time()
        self.mean_embeddings = [None] * self.max_speakers
        self.speaker_embeddings = [[] for _ in range(self.max_speakers)]
        self.last_similarity = 0.0
        self.active_speakers = set([0])
        
    def set_max_speakers(self, max_speakers):
        """Update the maximum number of speakers"""
        new_max = min(max_speakers, ABSOLUTE_MAX_SPEAKERS)
        
        if new_max < self.max_speakers:
            for speaker_id in list(self.active_speakers):
                if speaker_id >= new_max:
                    self.active_speakers.discard(speaker_id)
            
            if self.current_speaker >= new_max:
                self.current_speaker = 0
        
        if new_max > self.max_speakers:
            self.mean_embeddings.extend([None] * (new_max - self.max_speakers))
            self.speaker_embeddings.extend([[] for _ in range(new_max - self.max_speakers)])
        else:
            self.mean_embeddings = self.mean_embeddings[:new_max]
            self.speaker_embeddings = self.speaker_embeddings[:new_max]
        
        self.max_speakers = new_max
        
    def set_change_threshold(self, threshold):
        """Update the threshold for detecting speaker changes"""
        self.change_threshold = max(0.1, min(threshold, 0.99))
        
    def add_embedding(self, embedding, timestamp=None):
        """Add a new embedding and check if there's a speaker change"""
        current_time = timestamp or time.time()
        
        if not self.previous_embeddings:
            self.previous_embeddings.append(embedding)
            self.speaker_embeddings[self.current_speaker].append(embedding)
            if self.mean_embeddings[self.current_speaker] is None:
                self.mean_embeddings[self.current_speaker] = embedding.copy()
            return self.current_speaker, 1.0
        
        current_mean = self.mean_embeddings[self.current_speaker]
        if current_mean is not None:
            similarity = 1.0 - cosine(embedding, current_mean)
        else:
            similarity = 1.0 - cosine(embedding, self.previous_embeddings[-1])
        
        self.last_similarity = similarity
        
        time_since_last_change = current_time - self.last_change_time
        is_speaker_change = False
        
        if time_since_last_change >= MIN_SEGMENT_DURATION:
            if similarity < self.change_threshold:
                best_speaker = self.current_speaker
                best_similarity = similarity
                
                for speaker_id in range(self.max_speakers):
                    if speaker_id == self.current_speaker:
                        continue
                        
                    speaker_mean = self.mean_embeddings[speaker_id]
                    
                    if speaker_mean is not None:
                        speaker_similarity = 1.0 - cosine(embedding, speaker_mean)
                        
                        if speaker_similarity > best_similarity:
                            best_similarity = speaker_similarity
                            best_speaker = speaker_id
                
                if best_speaker != self.current_speaker:
                    is_speaker_change = True
                    self.current_speaker = best_speaker
                elif len(self.active_speakers) < self.max_speakers:
                    for new_id in range(self.max_speakers):
                        if new_id not in self.active_speakers:
                            is_speaker_change = True
                            self.current_speaker = new_id
                            self.active_speakers.add(new_id)
                            break
        
        if is_speaker_change:
            self.last_change_time = current_time
        
        self.previous_embeddings.append(embedding)
        if len(self.previous_embeddings) > EMBEDDING_HISTORY_SIZE:
            self.previous_embeddings.pop(0)
        
        self.speaker_embeddings[self.current_speaker].append(embedding)
        self.active_speakers.add(self.current_speaker)
        
        if len(self.speaker_embeddings[self.current_speaker]) > 30:
            self.speaker_embeddings[self.current_speaker] = self.speaker_embeddings[self.current_speaker][-30:]
            
        if self.speaker_embeddings[self.current_speaker]:
            self.mean_embeddings[self.current_speaker] = np.mean(
                self.speaker_embeddings[self.current_speaker], axis=0
            )
        
        return self.current_speaker, similarity
    
    def get_color_for_speaker(self, speaker_id):
        """Return color for speaker ID"""
        if 0 <= speaker_id < len(SPEAKER_COLORS):
            return SPEAKER_COLORS[speaker_id]
        return "#FFFFFF"
    
    def get_status_info(self):
        """Return status information about the speaker change detector"""
        speaker_counts = [len(self.speaker_embeddings[i]) for i in range(self.max_speakers)]
        
        return {
            "current_speaker": self.current_speaker,
            "speaker_counts": speaker_counts,
            "active_speakers": len(self.active_speakers),
            "max_speakers": self.max_speakers,
            "last_similarity": self.last_similarity,
            "threshold": self.change_threshold
        }


class RealtimeSpeakerDiarization:
    def __init__(self):
        self.encoder = None
        self.audio_processor = None
        self.speaker_detector = None
        self.recorder = None
        self.sentence_queue = queue.Queue()
        self.full_sentences = []
        self.sentence_speakers = []
        self.pending_sentences = []
        self.displayed_text = ""
        self.last_realtime_text = ""
        self.is_running = False
        self.change_threshold = DEFAULT_CHANGE_THRESHOLD
        self.max_speakers = DEFAULT_MAX_SPEAKERS
        self.current_conversation = ""
        
    def initialize_models(self):
        """Initialize the speaker encoder model"""
        try:
            device_str = "cuda" if torch.cuda.is_available() else "cpu"
            print(f"Using device: {device_str}")
            
            self.encoder = SpeechBrainEncoder(device=device_str)
            success = self.encoder.load_model()
            
            if success:
                self.audio_processor = AudioProcessor(self.encoder)
                self.speaker_detector = SpeakerChangeDetector(
                    embedding_dim=self.encoder.embedding_dim,
                    change_threshold=self.change_threshold,
                    max_speakers=self.max_speakers
                )
                print("ECAPA-TDNN model loaded successfully!")
                return True
            else:
                print("Failed to load ECAPA-TDNN model")
                return False
        except Exception as e:
            print(f"Model initialization error: {e}")
            return False
    
    def live_text_detected(self, text):
        """Callback for real-time transcription updates"""
        text = text.strip()
        if text:
            sentence_delimiters = '.?!。'
            prob_sentence_end = (
                len(self.last_realtime_text) > 0
                and text[-1] in sentence_delimiters
                and self.last_realtime_text[-1] in sentence_delimiters
            )

            self.last_realtime_text = text

            if prob_sentence_end and FAST_SENTENCE_END:
                self.recorder.stop()
            elif prob_sentence_end:
                self.recorder.post_speech_silence_duration = SILENCE_THRESHS[0]
            else:
                self.recorder.post_speech_silence_duration = SILENCE_THRESHS[1]
    
    def process_final_text(self, text):
        """Process final transcribed text with speaker embedding"""
        text = text.strip()
        if text:
            try:
                bytes_data = self.recorder.last_transcription_bytes
                self.sentence_queue.put((text, bytes_data))
                self.pending_sentences.append(text)
            except Exception as e:
                print(f"Error processing final text: {e}")
    
    def process_sentence_queue(self):
        """Process sentences in the queue for speaker detection"""
        while self.is_running:
            try:
                text, bytes_data = self.sentence_queue.get(timeout=1)
                
                # Convert audio data to int16
                audio_int16 = np.frombuffer(bytes_data, dtype=np.int16)
                
                # Extract speaker embedding
                speaker_embedding = self.audio_processor.extract_embedding(audio_int16)
                
                # Store sentence and embedding
                self.full_sentences.append((text, speaker_embedding))
                
                # Fill in missing speaker assignments
                while len(self.sentence_speakers) < len(self.full_sentences) - 1:
                    self.sentence_speakers.append(0)
                
                # Detect speaker changes
                speaker_id, similarity = self.speaker_detector.add_embedding(speaker_embedding)
                self.sentence_speakers.append(speaker_id)
                
                # Remove from pending
                if text in self.pending_sentences:
                    self.pending_sentences.remove(text)
                
                # Update conversation display
                self.current_conversation = self.get_formatted_conversation()
                    
            except queue.Empty:
                continue
            except Exception as e:
                print(f"Error processing sentence: {e}")
    
    def start_recording(self):
        """Start the recording and transcription process"""
        if self.encoder is None:
            return "Please initialize models first!"
        
        try:
            # Setup recorder configuration for WebRTC input
            recorder_config = {
                'spinner': False,
                'use_microphone': False,  # We'll feed audio manually
                'model': FINAL_TRANSCRIPTION_MODEL,
                'language': TRANSCRIPTION_LANGUAGE,
                'silero_sensitivity': SILERO_SENSITIVITY,
                'webrtc_sensitivity': WEBRTC_SENSITIVITY,
                'post_speech_silence_duration': SILENCE_THRESHS[1],
                'min_length_of_recording': MIN_LENGTH_OF_RECORDING,
                'pre_recording_buffer_duration': PRE_RECORDING_BUFFER_DURATION,
                'min_gap_between_recordings': 0,
                'enable_realtime_transcription': True,
                'realtime_processing_pause': 0,
                'realtime_model_type': REALTIME_TRANSCRIPTION_MODEL,
                'on_realtime_transcription_update': self.live_text_detected,
                'beam_size': FINAL_BEAM_SIZE,
                'beam_size_realtime': REALTIME_BEAM_SIZE,
                'buffer_size': BUFFER_SIZE,
                'sample_rate': SAMPLE_RATE,
            }

            self.recorder = AudioToTextRecorder(**recorder_config)
            
            # Start sentence processing thread
            self.is_running = True
            self.sentence_thread = threading.Thread(target=self.process_sentence_queue, daemon=True)
            self.sentence_thread.start()
            
            # Start transcription thread
            self.transcription_thread = threading.Thread(target=self.run_transcription, daemon=True)
            self.transcription_thread.start()
            
            return "Recording started successfully! FastRTC audio input ready."
            
        except Exception as e:
            return f"Error starting recording: {e}"
    
    def run_transcription(self):
        """Run the transcription loop"""
        try:
            while self.is_running:
                self.recorder.text(self.process_final_text)
        except Exception as e:
            print(f"Transcription error: {e}")
    
    def stop_recording(self):
        """Stop the recording process"""
        self.is_running = False
        if self.recorder:
            self.recorder.stop()
        return "Recording stopped!"
    
    def clear_conversation(self):
        """Clear all conversation data"""
        self.full_sentences = []
        self.sentence_speakers = []
        self.pending_sentences = []
        self.displayed_text = ""
        self.last_realtime_text = ""
        self.current_conversation = "Conversation cleared!"
        
        if self.speaker_detector:
            self.speaker_detector = SpeakerChangeDetector(
                embedding_dim=self.encoder.embedding_dim,
                change_threshold=self.change_threshold,
                max_speakers=self.max_speakers
            )
        
        return "Conversation cleared!"
    
    def update_settings(self, threshold, max_speakers):
        """Update speaker detection settings"""
        self.change_threshold = threshold
        self.max_speakers = max_speakers
        
        if self.speaker_detector:
            self.speaker_detector.set_change_threshold(threshold)
            self.speaker_detector.set_max_speakers(max_speakers)
        
        return f"Settings updated: Threshold={threshold:.2f}, Max Speakers={max_speakers}"
    
    def get_formatted_conversation(self):
        """Get the formatted conversation with speaker colors"""
        try:
            sentences_with_style = []
            
            # Process completed sentences
            for i, sentence in enumerate(self.full_sentences):
                sentence_text, _ = sentence
                if i >= len(self.sentence_speakers):
                    color = "#FFFFFF"
                    speaker_name = "Unknown"
                else:
                    speaker_id = self.sentence_speakers[i]
                    color = self.speaker_detector.get_color_for_speaker(speaker_id)
                    speaker_name = f"Speaker {speaker_id + 1}"
                    
                sentences_with_style.append(
                    f'<span style="color:{color};"><b>{speaker_name}:</b> {sentence_text}</span>')
            
            # Add pending sentences
            for pending_sentence in self.pending_sentences:
                sentences_with_style.append(
                    f'<span style="color:#60FFFF;"><b>Processing:</b> {pending_sentence}</span>')
            
            if sentences_with_style:
                return "<br><br>".join(sentences_with_style)
            else:
                return "Waiting for speech input..."
                
        except Exception as e:
            return f"Error formatting conversation: {e}"
    
    def get_status_info(self):
        """Get current status information"""
        if not self.speaker_detector:
            return "Speaker detector not initialized"
        
        try:
            status = self.speaker_detector.get_status_info()
            
            status_lines = [
                f"**Current Speaker:** {status['current_speaker'] + 1}",
                f"**Active Speakers:** {status['active_speakers']} of {status['max_speakers']}",
                f"**Last Similarity:** {status['last_similarity']:.3f}",
                f"**Change Threshold:** {status['threshold']:.2f}",
                f"**Total Sentences:** {len(self.full_sentences)}",
                "",
                "**Speaker Segment Counts:**"
            ]
            
            for i in range(status['max_speakers']):
                color_name = SPEAKER_COLOR_NAMES[i] if i < len(SPEAKER_COLOR_NAMES) else f"Speaker {i+1}"
                status_lines.append(f"Speaker {i+1} ({color_name}): {status['speaker_counts'][i]}")
            
            return "\n".join(status_lines)
            
        except Exception as e:
            return f"Error getting status: {e}"

    def process_audio(self, audio_data):
        """Process audio data from FastRTC"""
        if not self.is_running or not self.recorder:
            return
        
        try:
            # Extract audio data from FastRTC format (sample_rate, numpy_array)
            sample_rate, audio_array = audio_data
            
            # Convert to int16 format
            if audio_array.dtype != np.int16:
                audio_array = (audio_array * 32767).astype(np.int16)
            
            # Convert to bytes and feed to recorder
            audio_bytes = audio_array.tobytes()
            self.recorder.feed_audio(audio_bytes)
        except Exception as e:
            print(f"Error processing FastRTC audio: {e}")


# FastRTC Audio Handler
class DiarizationHandler(AsyncStreamHandler):
    def __init__(self, diarization_system):
        super().__init__()
        self.diarization_system = diarization_system
        
    def copy(self):
        # Return a fresh handler for each new stream connection
        return DiarizationHandler(self.diarization_system)
    
    async def emit(self):
        """Not used in this implementation"""
        return None
    
    async def receive(self, data):
        """Receive audio data from FastRTC and process it"""
        if self.diarization_system.is_running:
            self.diarization_system.process_audio(data)


# Global instance
diarization_system = RealtimeSpeakerDiarization()


def initialize_system():
    """Initialize the diarization system"""
    success = diarization_system.initialize_models()
    if success:
        return "✅ System initialized successfully! Models loaded."
    else:
        return "❌ Failed to initialize system. Please check the logs."


def start_recording():
    """Start recording and transcription"""
    return diarization_system.start_recording()


def stop_recording():
    """Stop recording and transcription"""
    return diarization_system.stop_recording()


def clear_conversation():
    """Clear the conversation"""
    return diarization_system.clear_conversation()


def update_settings(threshold, max_speakers):
    """Update system settings"""
    return diarization_system.update_settings(threshold, max_speakers)


def get_conversation():
    """Get the current conversation"""
    return diarization_system.get_formatted_conversation()


def get_status():
    """Get system status"""
    return diarization_system.get_status_info()


# Get Cloudflare TURN credentials for FastRTC
async def get_cloudflare_credentials():
    # Check if HF_TOKEN is set in environment
    hf_token = os.environ.get("HF_TOKEN")
    
    # If not set, use a default Hugging Face token if available
    if not hf_token:
        # Log a warning that user should set their own token
        print("Warning: HF_TOKEN environment variable not set. Please set your own Hugging Face token.")
        # Try to use the Hugging Face token from the environment
        from huggingface_hub import HfApi
        try:
            api = HfApi()
            hf_token = api.token
            if not hf_token:
                print("Error: No Hugging Face token available. TURN relay may not work properly.")
        except:
            print("Error: Failed to get Hugging Face token. TURN relay may not work properly.")
    
    # Get Cloudflare TURN credentials using the Hugging Face token
    if hf_token:
        try:
            return await get_cloudflare_turn_credentials_async(hf_token=hf_token)
        except Exception as e:
            print(f"Error getting Cloudflare TURN credentials: {e}")
    
    # Fallback to a default configuration that may not work
    return {
        "iceServers": [
            {
                "urls": "stun:stun.l.google.com:19302"
            }
        ]
    }


# Setup FastRTC stream handler with TURN server configuration
def setup_fastrtc_handler():
    """Set up FastRTC audio stream handler with TURN server configuration"""
    handler = DiarizationHandler(diarization_system)
    
    # Get server-side credentials (longer TTL)
    server_credentials = get_cloudflare_turn_credentials(ttl=360000)
    
    stream = Stream(
        handler=handler, 
        modality="audio", 
        mode="receive",
        rtc_configuration=get_cloudflare_credentials,  # Async function for client-side credentials
        server_rtc_configuration=server_credentials    # Server-side credentials with longer TTL
    )
    
    return stream


# Create Gradio interface
def create_interface():
    with gr.Blocks(title="Real-time Speaker Diarization", theme=gr.themes.Monochrome()) as interface:
        gr.Markdown("# 🎤 Real-time Speech Recognition with Speaker Diarization")
        gr.Markdown("This app performs real-time speech recognition with automatic speaker identification and color-coding.")
        
        with gr.Row():
            with gr.Column(scale=2):
                # FastRTC Audio Component
                fastrtc_html = gr.HTML("""
                <div class="fastrtc-container" style="margin-bottom: 20px;">
                    <h3>🎙️ FastRTC Audio Input</h3>
                    <p>Click the button below to start the audio stream:</p>
                    <button id="start-fastrtc" style="background: #3498db; color: white; padding: 10px 20px; border: none; border-radius: 5px; cursor: pointer;">
                        Start FastRTC Audio
                    </button>
                    <div id="fastrtc-status" style="margin-top: 10px; font-style: italic;">Not connected</div>
                    <script>
                        document.getElementById('start-fastrtc').addEventListener('click', function() {
                            document.getElementById('fastrtc-status').textContent = 'Connecting...';
                            // FastRTC will initialize the connection
                            fetch('/start-rtc', { method: 'POST' })
                                .then(response => response.text())
                                .then(data => {
                                    document.getElementById('fastrtc-status').textContent = 'Connected! Speak now...';
                                })
                                .catch(error => {
                                    document.getElementById('fastrtc-status').textContent = 'Connection error: ' + error;
                                });
                        });
                    </script>
                </div>
                """)
                
                # Main conversation display
                conversation_output = gr.HTML(
                    value="<i>Click 'Initialize System' to start...</i>",
                    label="Live Conversation"
                )
                
                # Control buttons
                with gr.Row():
                    init_btn = gr.Button("🔧 Initialize System", variant="secondary")
                    start_btn = gr.Button("🎙️ Start Recording", variant="primary", interactive=False)
                    stop_btn = gr.Button("⏹️ Stop Recording", variant="stop", interactive=False)
                    clear_btn = gr.Button("🗑️ Clear Conversation", interactive=False)
                
                # Status display
                status_output = gr.Textbox(
                    label="System Status",
                    value="System not initialized",
                    lines=8,
                    interactive=False
                )
            
            with gr.Column(scale=1):
                # Settings panel
                gr.Markdown("## ⚙️ Settings")
                
                threshold_slider = gr.Slider(
                    minimum=0.1,
                    maximum=0.95,
                    step=0.05,
                    value=DEFAULT_CHANGE_THRESHOLD,
                    label="Speaker Change Sensitivity",
                    info="Lower values = more sensitive to speaker changes"
                )
                
                max_speakers_slider = gr.Slider(
                    minimum=2,
                    maximum=ABSOLUTE_MAX_SPEAKERS,
                    step=1,
                    value=DEFAULT_MAX_SPEAKERS,
                    label="Maximum Number of Speakers"
                )
                
                update_settings_btn = gr.Button("Update Settings")
                
                # Instructions
                gr.Markdown("## 📝 Instructions")
                gr.Markdown("""
                1. Click **Initialize System** to load models
                2. Click **Start Recording** to begin processing
                3. Click **Start FastRTC Audio** to connect your microphone
                4. Allow microphone access when prompted
                5. Speak into your microphone
                6. Watch real-time transcription with speaker labels
                7. Adjust settings as needed
                """)
                
                # Speaker color legend
                gr.Markdown("## 🎨 Speaker Colors")
                color_info = []
                for i, (color, name) in enumerate(zip(SPEAKER_COLORS, SPEAKER_COLOR_NAMES)):
                    color_info.append(f'<span style="color:{color};">■</span> Speaker {i+1} ({name})')
                
                gr.HTML("<br>".join(color_info[:DEFAULT_MAX_SPEAKERS]))
                
                # FastRTC Integration Notice
                gr.Markdown("""
                ## ℹ️ About FastRTC
                This app uses FastRTC for low-latency audio streaming. 
                For optimal performance, use a modern browser and allow microphone access when prompted.
                """)
                
                # Hugging Face Token Information
                gr.Markdown("""
                ## 🔑 Hugging Face Token
                This app uses Cloudflare TURN server via Hugging Face integration.
                If audio connection fails, set your HF_TOKEN environment variable in the Space settings.
                """)
        
        # Auto-refresh conversation and status
        def refresh_display():
            return diarization_system.get_formatted_conversation(), diarization_system.get_status_info()
        
        # Event handlers
        def on_initialize():
            result = initialize_system()
            if "successfully" in result:
                return (
                    result,
                    gr.update(interactive=True),   # start_btn
                    gr.update(interactive=True),   # clear_btn
                    get_conversation(),
                    get_status()
                )
            else:
                return (
                    result,
                    gr.update(interactive=False),  # start_btn
                    gr.update(interactive=False),  # clear_btn
                    get_conversation(),
                    get_status()
                )
        
        def on_start():
            result = start_recording()
            return (
                result,
                gr.update(interactive=False),  # start_btn
                gr.update(interactive=True),   # stop_btn
            )
        
        def on_stop():
            result = stop_recording()
            return (
                result,
                gr.update(interactive=True),   # start_btn
                gr.update(interactive=False),  # stop_btn
            )
        
        # Connect event handlers
        init_btn.click(
            on_initialize,
            outputs=[status_output, start_btn, clear_btn, conversation_output, status_output]
        )
        
        start_btn.click(
            on_start,
            outputs=[status_output, start_btn, stop_btn]
        )
        
        stop_btn.click(
            on_stop,
            outputs=[status_output, start_btn, stop_btn]
        )
        
        clear_btn.click(
            clear_conversation,
            outputs=[status_output]
        )
        
        update_settings_btn.click(
            update_settings,
            inputs=[threshold_slider, max_speakers_slider],
            outputs=[status_output]
        )
        
        # Auto-refresh every 2 seconds when recording
        refresh_timer = gr.Timer(2.0)
        refresh_timer.tick(
            refresh_display,
            outputs=[conversation_output, status_output]
        )
    
    return interface


# 1) Create FastAPI app
app = FastAPI()

# 2) Create Gradio interface
gradio_interface = create_interface()

# 3) Mount Gradio onto FastAPI at root
app = gr.mount_gradio_app(app, gradio_interface, path="/")

# 4) Initialize and mount FastRTC stream on the same app
rtc_stream = setup_fastrtc_handler()
rtc_stream.mount(app)

# 5) Expose an endpoint to trigger the client-side RTC handshake
@app.post("/start-rtc")
async def start_rtc():
    await rtc_stream.start_client()
    return {"status": "success"}

# 6) Local dev via uvicorn; HF Spaces will auto-detect 'app' and ignore this
if __name__ == "__main__":
    uvicorn.run(app, host="0.0.0.0", port=7860)