diff --git a/voicebot/bots/vibevoice.py b/voicebot/bots/vibevoice.py
index 89ab6ae..6c397f5 100644
--- a/voicebot/bots/vibevoice.py
+++ b/voicebot/bots/vibevoice.py
@@ -249,24 +249,37 @@ class WaveformVideoTrack(MediaStreamTrack):
             # Non-critical overlay; ignore failures
             pass
 
-        # Select the most active audio buffer and get its speech status
+        # Prefer the generated TTS audio buffer if present. vibevoice ignores
+        # incoming WebRTC audio; the global `_audio_data` is populated by the
+        # TTS background worker and represents the audio we want to visualize.
         best_proc = None
         best_rms = 0.0
         speech_info = None
 
         try:
-            for pname, arr in self.__class__.buffer.items():
+            if _audio_data is not None and getattr(_audio_data, "size", 0) > 0:
+                # Use a synthetic pname to indicate TTS-generated audio and
+                # copy the buffer for safe local use.
                 try:
-                    if len(arr) == 0:
-                        rms = 0.0
-                    else:
-                        rms = float(np.sqrt(np.mean(arr**2)))
-                    if rms > best_rms:
-                        best_rms = rms
-                        best_proc = (pname, arr.copy())
-                        speech_info = self.__class__.speech_status.get(pname, {})
+                    tts_arr = np.asarray(_audio_data, dtype=np.float32)
+                    best_proc = ("__tts__", tts_arr.copy())
+                    # Mark as speech for coloring purposes
+                    speech_info = {"is_speech": True, "energy_check": True}
                 except Exception:
-                    continue
+                    best_proc = None
+            else:
+                for pname, arr in self.__class__.buffer.items():
+                    try:
+                        if len(arr) == 0:
+                            rms = 0.0
+                        else:
+                            rms = float(np.sqrt(np.mean(arr**2)))
+                        if rms > best_rms:
+                            best_rms = rms
+                            best_proc = (pname, arr.copy())
+                            speech_info = self.__class__.speech_status.get(pname, {})
+                    except Exception:
+                        continue
         except Exception:
             best_proc = None
 
@@ -509,6 +522,14 @@ async def handle_track_received(peer: Peer, track: MediaStreamTrack) -> None:
         logger.info(f"Ignoring non-audio track from {peer.peer_name}: {track.kind}")
         return
 
+    # This bot (vibevoice) does not use incoming WebRTC audio for TTS or
+    # waveform rendering. Ignore audio tracks entirely to avoid populating
+    # the shared waveform buffer with remote audio. The generated TTS audio
+    # is stored in the module-global `_audio_data` and will be used for the
+    # waveform and silent audio track playback instead.
+    logger.info(f"vibevoice: ignoring incoming audio track from {peer.peer_name}")
+    return
+
     # Initialize raw audio buffer for immediate graphing
     if peer.peer_name not in WaveformVideoTrack.buffer:
         WaveformVideoTrack.buffer[peer.peer_name] = np.array([], dtype=np.float32)
diff --git a/voicebot/bots/vibevoicetts.py b/voicebot/bots/vibevoicetts.py
new file mode 100644
index 0000000..8727ac1
--- /dev/null
+++ b/voicebot/bots/vibevoicetts.py
@@ -0,0 +1,490 @@
+import os
+import re
+import traceback
+from typing import Any, List, Tuple, Union, Optional
+import time
+import torch
+import numpy as np
+
+from vibevoice.modular.modeling_vibevoice_inference import VibeVoiceForConditionalGenerationInference
+from vibevoice.processor.vibevoice_processor import VibeVoiceProcessor
+
+from shared.logger import logger
+
+class VoiceMapper:
+    """Maps speaker names to voice file paths"""
+    
+    def __init__(self, voices_dir: Optional[str] = None):
+        if voices_dir is None:
+            voices_dir = os.path.join(os.path.dirname(__file__), "voices")
+        self.voices_dir = voices_dir
+        self.setup_voice_presets()
+
+        # Change name according to our preset wav file
+        new_dict = {}
+        for name, path in self.voice_presets.items():
+            if '_' in name:
+                name = name.split('_')[0]
+            if '-' in name:
+                name = name.split('-')[-1]
+            new_dict[name] = path
+        self.voice_presets.update(new_dict)
+
+    def setup_voice_presets(self):
+        """Setup voice presets by scanning the voices directory."""
+        # Check if voices directory exists
+        if not os.path.exists(self.voices_dir):
+            logger.info(f"Warning: Voices directory not found at {self.voices_dir}")
+            self.voice_presets = {}
+            self.available_voices = {}
+            return
+        
+        # Scan for all WAV files in the voices directory
+        self.voice_presets = {}
+        
+        # Get all .wav files in the voices directory
+        wav_files = [f for f in os.listdir(self.voices_dir) 
+                    if f.lower().endswith('.wav') and os.path.isfile(os.path.join(self.voices_dir, f))]
+        
+        # Create dictionary with filename (without extension) as key
+        for wav_file in wav_files:
+            # Remove .wav extension to get the name
+            name = os.path.splitext(wav_file)[0]
+            # Create full path
+            full_path = os.path.join(self.voices_dir, wav_file)
+            self.voice_presets[name] = full_path
+        
+        # Sort the voice presets alphabetically by name for better UI
+        self.voice_presets = dict(sorted(self.voice_presets.items()))
+        
+        # Filter out voices that don't exist (this is now redundant but kept for safety)
+        self.available_voices = {
+            name: path for name, path in self.voice_presets.items()
+            if os.path.exists(path)
+        }
+        
+        logger.info(f"Found {len(self.available_voices)} voice files in {self.voices_dir}")
+        if self.available_voices:
+            logger.info(f"Available voices: {', '.join(self.available_voices.keys())}")
+
+    def get_voice_path(self, speaker_name: str) -> str:
+        """Get voice file path for a given speaker name"""
+        # First try exact match
+        if speaker_name in self.voice_presets:
+            return self.voice_presets[speaker_name]
+        
+        # Try partial matching (case insensitive)
+        speaker_lower = speaker_name.lower()
+        for preset_name, path in self.voice_presets.items():
+            if preset_name.lower() in speaker_lower or speaker_lower in preset_name.lower():
+                return path
+        
+        # Default to first voice if no match found
+        if not self.voice_presets:
+            raise ValueError("No voice files available")
+        
+        default_voice = list(self.voice_presets.values())[0]
+        logger.info(f"Warning: No voice preset found for '{speaker_name}', using default voice: {default_voice}")
+        return default_voice
+
+
+class VibeVoiceTTS:
+    """
+    A reusable Text-to-Speech engine using VibeVoice model.
+    
+    Example usage:
+        tts_engine = VibeVoiceTTS(model_path="microsoft/VibeVoice-1.5b")
+        audio_data = tts_engine.text_to_speech(
+            text="Speaker 1: Hello world!\nSpeaker 2: How are you?",
+            speaker_names=["Andrew", "Ava"]
+        )
+    """
+    
+    def __init__(
+        self, 
+        model_path: str = "microsoft/VibeVoice-1.5b",
+        device: Optional[str] = None,
+        voices_dir: Optional[str] = None,
+        cfg_scale: float = 1.3,
+        ddpm_steps: int = 10
+    ):
+        """
+        Initialize the TTS engine with model and configuration.
+        
+        Args:
+            model_path: Path to the HuggingFace model directory
+            device: Device for inference ('cuda', 'mps', 'cpu'). Auto-detected if None
+            voices_dir: Directory containing voice sample .wav files
+            cfg_scale: CFG (Classifier-Free Guidance) scale for generation
+            ddpm_steps: Number of DDPM inference steps
+        """
+        self.model_path = model_path
+        self.cfg_scale = cfg_scale
+        self.ddpm_steps = ddpm_steps
+        
+        # Auto-detect device if not specified
+        if device is None:
+            if torch.xpu.is_available():
+                device = "xpu"
+            elif torch.cuda.is_available():
+                device = "cuda"
+            elif torch.backends.mps.is_available():
+                device = "mps"
+            else:
+                device = "cpu"
+        
+        # Handle potential typos
+        if device.lower() == "mpx":
+            logger.info("Note: device 'mpx' detected, treating it as 'mps'.")
+            device = "mps"
+        
+        # Validate mps availability
+        if device == "mps" and not torch.backends.mps.is_available():
+            logger.info("Warning: MPS not available. Falling back to CPU.")
+            device = "cpu"
+        
+        self.device = device
+        logger.info(f"Using device: {self.device}")
+        
+        # Initialize voice mapper
+        self.voice_mapper = VoiceMapper(voices_dir)
+        
+        # Load model and processor
+        self._load_model()
+        
+    def _load_model(self):
+        """Load the model and processor with device-specific configuration."""
+        logger.info(f"Loading processor & model from {self.model_path}")
+        self.processor = VibeVoiceProcessor.from_pretrained(self.model_path)
+        
+        # Decide dtype & attention implementation
+        if self.device == "mps":
+            load_dtype = torch.float32  # MPS requires float32
+            attn_impl_primary = "sdpa"  # flash_attention_2 not supported on MPS
+        elif self.device == "cuda":
+            load_dtype = torch.bfloat16
+            attn_impl_primary = "flash_attention_2"
+        elif self.device == "xpu":
+            load_dtype = torch.bfloat16
+            attn_impl_primary = "sdpa"  # flash_attention_2 not supported on XPU
+        else:  # cpu
+            load_dtype = torch.float32
+            attn_impl_primary = "sdpa"
+        
+        logger.info(f"Using torch_dtype: {load_dtype}, attn_implementation: {attn_impl_primary}")
+        
+        # Load model with device-specific logic
+        try:
+            if self.device == "mps":
+                self.model = VibeVoiceForConditionalGenerationInference.from_pretrained(
+                    self.model_path,
+                    torch_dtype=load_dtype,
+                    attn_implementation=attn_impl_primary,
+                    device_map=None,  # load then move
+                )
+                self.model.to("mps")
+            elif self.device == "cuda":
+                self.model = VibeVoiceForConditionalGenerationInference.from_pretrained(
+                    self.model_path,
+                    torch_dtype=load_dtype,
+                    device_map="cuda",
+                    attn_implementation=attn_impl_primary,
+                )
+            elif self.device == "xpu":
+                self.model = VibeVoiceForConditionalGenerationInference.from_pretrained(
+                    self.model_path,
+                    torch_dtype=load_dtype,
+                    device_map="xpu",
+                    attn_implementation=attn_impl_primary,
+                )
+            else:  # cpu
+                self.model = VibeVoiceForConditionalGenerationInference.from_pretrained(
+                    self.model_path,
+                    torch_dtype=load_dtype,
+                    device_map="cpu",
+                    attn_implementation=attn_impl_primary,
+                )
+        except Exception as e:
+            if attn_impl_primary == 'flash_attention_2':
+                logger.info(f"[ERROR] : {type(e).__name__}: {e}")
+                logger.info(traceback.format_exc())
+                logger.info("Error loading the model. Trying to use SDPA. However, note that only flash_attention_2 has been fully tested, and using SDPA may result in lower audio quality.")
+                self.model = VibeVoiceForConditionalGenerationInference.from_pretrained(
+                    self.model_path,
+                    torch_dtype=load_dtype,
+                    device_map=(self.device if self.device in ("cuda", "cpu", "xpu") else None),
+                    attn_implementation='sdpa'
+                )
+                if self.device == "mps":
+                    self.model.to("mps")
+            else:
+                raise e
+        
+        self.model.eval()
+        self.model.set_ddpm_inference_steps(num_steps=self.ddpm_steps)
+        
+        if hasattr(self.model.model, 'language_model'):
+            logger.info(f"Language model attention: {self.model.model.language_model.config._attn_implementation}")
+        
+        logger.info("Model loaded successfully!")
+
+    def _parse_script(self, text: str) -> Tuple[List[str], List[str]]:
+        """
+        Parse script text and extract speakers and their text.
+        Supports format: "Speaker 1: text", "Speaker 2: text", etc.
+        
+        Returns: (scripts, speaker_numbers)
+        """
+        lines = text.strip().split('\n')
+        scripts = []
+        speaker_numbers = []
+        
+        # Pattern to match "Speaker X:" format where X is a number
+        speaker_pattern = r'^Speaker\s+(\d+):\s*(.*)$'
+        
+        current_speaker = None
+        current_text = ""
+        
+        for line in lines:
+            line = line.strip()
+            if not line:
+                continue
+                
+            match = re.match(speaker_pattern, line, re.IGNORECASE)
+            if match:
+                # If we have accumulated text from previous speaker, save it
+                if current_speaker and current_text:
+                    scripts.append(f"Speaker {current_speaker}: {current_text.strip()}")
+                    speaker_numbers.append(current_speaker)
+                
+                # Start new speaker
+                current_speaker = match.group(1).strip()
+                current_text = match.group(2).strip()
+            else:
+                # Continue text for current speaker
+                if current_text:
+                    current_text += " " + line
+                else:
+                    current_text = line
+        
+        # Don't forget the last speaker
+        if current_speaker and current_text:
+            scripts.append(f"Speaker {current_speaker}: {current_text.strip()}")
+            speaker_numbers.append(current_speaker)
+        
+        return scripts, speaker_numbers
+
+    def text_to_speech(
+        self, 
+        text: str, 
+        speaker_names: Union[str, List[str]] = None,
+        cfg_scale: Optional[float] = None,
+        verbose: bool = False
+    ) -> np.ndarray:
+        """
+        Convert text to speech and return audio data.
+        
+        Args:
+            text: Input text with speaker labels (e.g., "Speaker 1: Hello\nSpeaker 2: Hi there")
+            speaker_names: Speaker name(s) to map to voice files. Can be single string or list.
+            cfg_scale: Override default CFG scale for this generation
+            verbose: Print detailed generation info
+            
+        Returns:
+            numpy.ndarray: Audio data as floating point array (sample rate: 24kHz)
+        """
+        if cfg_scale is None:
+            cfg_scale = self.cfg_scale
+            
+        # Parse the script to get speaker segments
+        scripts, speaker_numbers = self._parse_script(text)
+        
+        if not scripts:
+            raise ValueError("No valid speaker scripts found in the input text")
+        
+        if verbose:
+            logger.info(f"Found {len(scripts)} speaker segments:")
+            for i, (script, speaker_num) in enumerate(zip(scripts, speaker_numbers)):
+                logger.info(f"  {i+1}. Speaker {speaker_num}")
+                logger.info(f"     Text preview: {script[:100]}...")
+        
+        # Handle speaker names
+        if speaker_names is None:
+            speaker_names = ["Andrew"]  # Default speaker
+        elif isinstance(speaker_names, str):
+            speaker_names = [speaker_names]
+        
+        # Map speaker numbers to provided speaker names
+        speaker_name_mapping = {}
+        for i, name in enumerate(speaker_names, 1):
+            speaker_name_mapping[str(i)] = name
+        
+        if verbose:
+            logger.info("\nSpeaker mapping:")
+            for speaker_num in set(speaker_numbers):
+                mapped_name = speaker_name_mapping.get(speaker_num, f"Speaker {speaker_num}")
+                logger.info(f"  Speaker {speaker_num} -> {mapped_name}")
+        
+        # Map speakers to voice files
+        voice_samples = []
+        actual_speakers = []
+        
+        # Get unique speaker numbers in order of first appearance
+        unique_speaker_numbers = []
+        seen = set()
+        for speaker_num in speaker_numbers:
+            if speaker_num not in seen:
+                unique_speaker_numbers.append(speaker_num)
+                seen.add(speaker_num)
+        
+        for speaker_num in unique_speaker_numbers:
+            speaker_name = speaker_name_mapping.get(speaker_num, f"Speaker {speaker_num}")
+            voice_path = self.voice_mapper.get_voice_path(speaker_name)
+            voice_samples.append(voice_path)
+            actual_speakers.append(speaker_name)
+            if verbose:
+                logger.info(f"Speaker {speaker_num} ('{speaker_name}') -> Voice: {os.path.basename(voice_path)}")
+        
+        # Prepare data for model
+        full_script = '\n'.join(scripts)
+        full_script = full_script.replace("'", "'")
+        
+        # Prepare inputs for the model
+        inputs = self.processor(
+            text=[full_script],  # Wrap in list for batch processing
+            voice_samples=[voice_samples],  # Wrap in list for batch processing
+            padding=True,
+            return_tensors="pt",
+            return_attention_mask=True,
+        )
+        
+        # Move tensors to target device
+        target_device = self.device if self.device != "cpu" else "cpu"
+        for k, v in inputs.items():
+            if torch.is_tensor(v):
+                inputs[k] = v.to(target_device)
+        
+        if verbose:
+            logger.info(f"Starting generation with cfg_scale: {cfg_scale}")
+        
+        # Generate audio
+        start_time = time.time()
+        outputs = self.model.generate(
+            **inputs,
+            max_new_tokens=None,
+            cfg_scale=cfg_scale,
+            tokenizer=self.processor.tokenizer,
+            generation_config={'do_sample': False},
+            verbose=verbose,
+        )
+        generation_time = time.time() - start_time
+        
+        if verbose:
+            logger.info(f"Generation time: {generation_time:.2f} seconds")
+            
+            # Calculate metrics
+            if outputs.speech_outputs and outputs.speech_outputs[0] is not None:
+                sample_rate = 24000
+                audio_samples = outputs.speech_outputs[0].shape[-1] if len(outputs.speech_outputs[0].shape) > 0 else len(outputs.speech_outputs[0])
+                audio_duration = audio_samples / sample_rate
+                rtf = generation_time / audio_duration if audio_duration > 0 else float('inf')
+                
+                logger.info(f"Generated audio duration: {audio_duration:.2f} seconds")
+                logger.info(f"RTF (Real Time Factor): {rtf:.2f}x")
+                
+                # Token metrics
+                input_tokens = inputs['input_ids'].shape[1]
+                output_tokens = outputs.sequences.shape[1]
+                generated_tokens = output_tokens - input_tokens
+                
+                logger.info(f"Prefilling tokens: {input_tokens}")
+                logger.info(f"Generated tokens: {generated_tokens}")
+                logger.info(f"Total tokens: {output_tokens}")
+        
+        # Return audio data as numpy array
+        if outputs.speech_outputs and outputs.speech_outputs[0] is not None:
+            audio_tensor = outputs.speech_outputs[0]
+            
+            # Convert to numpy array on CPU
+            if hasattr(audio_tensor, 'cpu'):
+                audio_data = audio_tensor.cpu().numpy()
+            else:
+                audio_data = np.array(audio_tensor)
+            
+            # Ensure it's a 1D array
+            if audio_data.ndim > 1:
+                audio_data = audio_data.squeeze()
+            
+            return audio_data
+        else:
+            raise RuntimeError("No audio output generated")
+
+    def get_available_voices(self) -> List[str]:
+        """Get list of available voice names."""
+        return list(self.voice_mapper.available_voices.keys())
+
+    def get_sample_rate(self) -> int:
+        """Get the sample rate of generated audio."""
+        return 24000  # VibeVoice uses 24kHz
+
+
+# Global instance for easy access
+_global_tts_engine = None
+
+def get_tts_engine(**kwargs) -> VibeVoiceTTS:
+    """
+    Get or create a global TTS engine instance.
+    
+    Args:
+        **kwargs: Arguments to pass to VibeVoiceTTS constructor (only used on first call)
+        
+    Returns:
+        VibeVoiceTTS: Global TTS engine instance
+    """
+    global _global_tts_engine
+    if _global_tts_engine is None:
+        _global_tts_engine = VibeVoiceTTS(**kwargs)
+    return _global_tts_engine
+
+
+# Convenience function for quick TTS
+def text_to_speech(text: str, speaker_names: Optional[Union[str, List[str]]] = None, **kwargs: dict[str, Any]) -> np.ndarray:
+    """
+    Quick text-to-speech conversion using global engine.
+    
+    Args:
+        text: Input text with speaker labels
+        speaker_names: Speaker name(s) to use
+        **kwargs: Additional arguments for TTS engine or text_to_speech method
+        
+    Returns:
+        numpy.ndarray: Audio data
+    """
+    # Separate engine kwargs from TTS kwargs
+    engine_kwargs = {k: v for k, v in kwargs.items() 
+                    if k in ['model_path', 'device', 'voices_dir', 'cfg_scale', 'ddpm_steps']}
+    tts_kwargs = {k: v for k, v in kwargs.items() if k not in engine_kwargs}
+    
+    engine = get_tts_engine(**engine_kwargs)
+    return engine.text_to_speech(text, speaker_names, **tts_kwargs)
+
+
+
+# Example usage:
+# Method 1: Create instance
+# tts_engine = VibeVoiceTTS(model_path="microsoft/VibeVoice-1.5b")
+# audio_data = tts_engine.text_to_speech(
+#     "Speaker 1: Hello world!\nSpeaker 2: How are you?", 
+#     speaker_names=["Andrew", "Ava"]
+# )
+
+# # Method 2: Use global instance
+# audio_data = text_to_speech(
+#     "Speaker 1: Hello world!", 
+#     speaker_names="Andrew",
+#     verbose=True
+# )
+
+# # Method 3: Global engine with custom config
+# engine = get_tts_engine(device="cuda", cfg_scale=1.5)
+# audio_data = engine.text_to_speech("Speaker 1: Hello!", ["Andrew"])
\ No newline at end of file
diff --git a/voicebot/requirements.txt b/voicebot/requirements.txt
index a3b3d60..f754094 100644
--- a/voicebot/requirements.txt
+++ b/voicebot/requirements.txt
@@ -1,6 +1,4 @@
 about-time==4.2.1
-absl-py==2.3.1
-accelerate==1.6.0
 aiofiles==24.1.0
 aiohappyeyeballs==2.6.1
 aiohttp==3.12.15
@@ -27,7 +25,6 @@ cycler==0.12.1
 datasets==4.1.0
 decorator==5.2.1
 deprecated==1.2.18
-diffusers==0.35.1
 dill==0.4.0
 distro==1.9.0
 dnspython==2.8.0
@@ -50,7 +47,6 @@ httpx==0.28.1
 huggingface-hub==0.34.5
 idna==3.10
 ifaddr==0.2.0
-importlib-metadata==8.7.0
 iniconfig==2.1.0
 jinja2==3.1.6
 jiter==0.11.0
@@ -66,7 +62,6 @@ markdown-it-py==4.0.0
 markupsafe==3.0.2
 matplotlib==3.10.6
 mdurl==0.1.2
-ml-collections==1.1.0
 ml-dtypes==0.5.3
 more-itertools==10.8.0
 mpmath==1.3.0
@@ -106,7 +101,6 @@ optimum-intel @ git+https://github.com/huggingface/optimum-intel.git@b9c151fec6b
 orjson==3.11.3
 packaging==25.0
 pandas==2.3.2
-peft==0.17.1
 pillow==11.3.0
 platformdirs==4.4.0
 pluggy==1.6.0
@@ -174,10 +168,8 @@ typing-inspection==0.4.1
 tzdata==2025.2
 urllib3==2.5.0
 uvicorn==0.35.0
--e file:///voicebot/VibeVoice
 watchdog==6.0.0
 websockets==15.0.1
 wrapt==1.17.3
 xxhash==3.5.0
 yarl==1.20.1
-zipp==3.23.0