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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
import json
import io
import wave
from fastrtc import Stream, ReplyOnPause, AsyncStreamHandler, get_stt_model
# Simplified configuration parameters
SILENCE_THRESHS = [0, 0.4]
FINAL_TRANSCRIPTION_MODEL = "moonshine/base" # Using FastRTC's moonshine model
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 DiarizationStreamHandler(AsyncStreamHandler):
"""FastRTC stream handler for real-time diarization"""
def __init__(self, diarization_system):
super().__init__(input_sample_rate=16000)
self.diarization_system = diarization_system
self.stt_model = get_stt_model(model=FINAL_TRANSCRIPTION_MODEL)
self.current_text = ""
self.current_audio_buffer = []
self.transcript_queue = queue.Queue()
def copy(self):
return DiarizationStreamHandler(self.diarization_system)
async def start_up(self):
"""Initialize the stream handler"""
pass
async def receive(self, frame):
"""Process incoming audio frame"""
# Extract audio data
sample_rate, audio_data = frame
# Convert to numpy array if needed
if isinstance(audio_data, torch.Tensor):
audio_data = audio_data.numpy()
# Add to buffer
self.current_audio_buffer.append(audio_data)
# If buffer is large enough, process it
if len(self.current_audio_buffer) > 3: # Process ~1.5 seconds of audio
# Concatenate audio data
combined_audio = np.concatenate(self.current_audio_buffer)
# Run speech-to-text
text = self.stt_model.stt((16000, combined_audio))
if text and text.strip():
# Save text and audio for processing
self.transcript_queue.put((text, combined_audio))
self.current_text = text
# Reset buffer but keep some overlap
if len(self.current_audio_buffer) > 5:
self.current_audio_buffer = self.current_audio_buffer[-2:]
async def emit(self):
"""Emit processed data"""
# Return current text as dummy; actual processing is done in background
return self.current_text
class RealtimeSpeakerDiarization:
def __init__(self):
self.encoder = None
self.audio_processor = None
self.speaker_detector = None
self.stream = None
self.stream_handler = 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
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 start_stream(self, app):
"""Start the FastRTC stream"""
if self.encoder is None:
return "Please initialize models first!"
try:
# Create a FastRTC stream handler
self.stream_handler = DiarizationStreamHandler(self)
# Create FastRTC stream
self.stream = Stream(
handler=self.stream_handler,
modality="audio",
mode="send-receive"
)
# Mount the stream to the provided FastAPI app
self.stream.mount(app)
# 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 diarization processor thread
self.diarization_thread = threading.Thread(target=self.process_transcript_queue, daemon=True)
self.diarization_thread.start()
return "Stream started successfully! Ready for audio input."
except Exception as e:
return f"Error starting stream: {e}"
def process_transcript_queue(self):
"""Process transcripts from the stream handler"""
while self.is_running:
try:
if self.stream_handler and not self.stream_handler.transcript_queue.empty():
text, audio_data = self.stream_handler.transcript_queue.get(timeout=1)
# Add to sentence queue for diarization
self.pending_sentences.append(text)
self.sentence_queue.put((text, audio_data))
except queue.Empty:
time.sleep(0.1) # Short sleep to prevent CPU hogging
except Exception as e:
print(f"Error processing transcript queue: {e}")
time.sleep(0.5) # Slightly longer sleep on error
def process_sentence_queue(self):
"""Process sentences in the queue for speaker detection"""
while self.is_running:
try:
text, audio_data = self.sentence_queue.get(timeout=1)
# Convert audio data to int16
if isinstance(audio_data, np.ndarray):
if audio_data.dtype != np.int16:
audio_int16 = (audio_data * 32767).astype(np.int16)
else:
audio_int16 = audio_data
else:
audio_int16 = np.int16(audio_data * 32767)
# 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)
except queue.Empty:
continue
except Exception as e:
print(f"Error processing sentence: {e}")
def stop_stream(self):
"""Stop the stream and processing"""
self.is_running = False
return "Stream 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 = ""
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"
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}"
# Global instance
diarization_system = RealtimeSpeakerDiarization()
# Create Gradio interface with FastAPI app integrated
def create_interface():
app = gr.Blocks(title="Real-time Speaker Diarization", theme=gr.themes.Monochrome())
with app:
gr.Markdown("# π€ Real-time Speech Recognition with Speaker Diarization")
gr.Markdown("This app performs real-time speech recognition with automatic speaker identification and color-coding using FastRTC.")
with gr.Row():
with gr.Column(scale=2):
# Main conversation display
conversation_output = gr.HTML(
value="<i>Click 'Initialize System' and then 'Start Stream' to begin...</i>",
label="Live Conversation"
)
# FastRTC microphone widget for visualization only (the real audio comes through FastRTC stream)
audio_widget = gr.Audio(
label="ποΈ Microphone Input (Click Start Stream to enable)",
type="microphone"
)
# Control buttons
with gr.Row():
init_btn = gr.Button("π§ Initialize System", variant="secondary")
start_btn = gr.Button("ποΈ Start Stream", variant="primary", interactive=False)
stop_btn = gr.Button("βΉοΈ Stop Stream", variant="stop", interactive=False)
clear_btn = gr.Button("ποΈ 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("## βοΈ 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("## π Instructions")
gr.Markdown("""
1. Click **Initialize System** to load models
2. Click **Start Stream** to begin processing
3. Allow microphone access when prompted
4. Speak into your microphone
5. Watch real-time transcription with speaker labels
6. Adjust settings as needed
""")
# QR code for mobile access
gr.Markdown("## π± Mobile Access")
gr.Markdown("Scan this QR code to access from mobile device:")
qr_code = gr.HTML("""
<div id="qrcode" style="text-align: center;"></div>
<script src="https://cdn.jsdelivr.net/npm/[email protected]/qrcode.min.js"></script>
<script>
setTimeout(function() {
var currentUrl = window.location.href;
var qr = qrcode(0, 'M');
qr.addData(currentUrl);
qr.make();
document.getElementById('qrcode').innerHTML = qr.createImgTag(5);
}, 1000);
</script>
""")
# Speaker color legend
gr.Markdown("## π¨ Speaker Colors")
color_info = []
for i, (color, name) in enumerate(zip(SPEAKER_COLORS, SPEAKER_COLOR_NAMES)):
color_info.append(f'<span style="color:{color};">β </span> Speaker {i+1} ({name})')
gr.HTML("<br>".join(color_info[:DEFAULT_MAX_SPEAKERS]))
# Auto-refresh conversation and status
def refresh_display():
return get_formatted_conversation(), get_status()
# 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_formatted_conversation(),
get_status()
)
else:
return (
result,
gr.update(interactive=False), # start_btn
gr.update(interactive=False), # clear_btn
get_formatted_conversation(),
get_status()
)
def on_start_stream():
result = start_stream(app)
return (
result,
gr.update(interactive=False), # start_btn
gr.update(interactive=True), # stop_btn
)
def on_stop_stream():
result = stop_stream()
return (
result,
gr.update(interactive=True), # start_btn
gr.update(interactive=False), # stop_btn
)
def initialize_system():
"""Initialize the diarization system"""
success = diarization_system.initialize_models()
if success:
return "β
System initialized successfully! Models loaded."
else:
return "β Failed to initialize system. Please check the logs."
def start_stream(app):
"""Start the FastRTC stream"""
return diarization_system.start_stream(app)
def stop_stream():
"""Stop the FastRTC stream"""
return diarization_system.stop_stream()
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_formatted_conversation():
"""Get the current conversation"""
return diarization_system.get_formatted_conversation()
def get_status():
"""Get system status"""
return diarization_system.get_status_info()
# Connect event handlers
init_btn.click(
on_initialize,
outputs=[status_output, start_btn, clear_btn, conversation_output, status_output]
)
start_btn.click(
on_start_stream,
outputs=[status_output, start_btn, stop_btn]
)
stop_btn.click(
on_stop_stream,
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 streaming
refresh_timer = gr.Timer(2.0)
refresh_timer.tick(
refresh_display,
outputs=[conversation_output, status_output]
)
return app
if __name__ == "__main__":
app = create_interface()
app.launch(
server_name="0.0.0.0",
server_port=7860,
share=True
)
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