diff --git a/api/src/core/config.py b/api/src/core/config.py
index d2e369b..d361a5c 100644
--- a/api/src/core/config.py
+++ b/api/src/core/config.py
@@ -28,8 +28,11 @@ class Settings(BaseSettings):
     target_min_tokens: int = 175  # Target minimum tokens per chunk
     target_max_tokens: int = 250  # Target maximum tokens per chunk
     absolute_max_tokens: int = 450  # Absolute maximum tokens per chunk
+    advanced_text_normalization: bool = True # Preproesses the text before misiki which leads 
 
-    gap_trim_ms: int = 250  # Amount to trim from streaming chunk ends in milliseconds
+    gap_trim_ms: int = 1  # Base amount to trim from streaming chunk ends in milliseconds
+    dynamic_gap_trim_padding_ms: int = 410 # Padding to add to dynamic gap trim
+    dynamic_gap_trim_padding_char_multiplier: dict[str,float] = {".":1,"!":0.9,"?":1,",":0.8}
 
     # Web Player Settings
     enable_web_player: bool = True  # Whether to serve the web player UI
diff --git a/api/src/inference/kokoro_v1.py b/api/src/inference/kokoro_v1.py
index 9f5e206..248593e 100644
--- a/api/src/inference/kokoro_v1.py
+++ b/api/src/inference/kokoro_v1.py
@@ -144,7 +144,7 @@ class KokoroV1(BaseModelBackend):
             pipeline = self._get_pipeline(pipeline_lang_code)
 
             logger.debug(
-                f"Generating audio from tokens with lang_code '{pipeline_lang_code}': '{tokens[:100]}...'"
+                f"Generating audio from tokens with lang_code '{pipeline_lang_code}': '{tokens[:100]}{'...' if len(tokens) > 100 else ''}'"
             )
             for result in pipeline.generate_from_tokens(
                 tokens=tokens, voice=voice_path, speed=speed, model=self._model
@@ -192,7 +192,6 @@ class KokoroV1(BaseModelBackend):
         """
         if not self.is_loaded:
             raise RuntimeError("Model not loaded")
-
         try:
             # Memory management for GPU
             if self._device == "cuda":
@@ -237,7 +236,7 @@ class KokoroV1(BaseModelBackend):
             pipeline = self._get_pipeline(pipeline_lang_code)
 
             logger.debug(
-                f"Generating audio for text with lang_code '{pipeline_lang_code}': '{text[:100]}...'"
+                f"Generating audio for text with lang_code '{pipeline_lang_code}': '{text[:100]}{'...' if len(text) > 100 else ''}'"
             )
             for result in pipeline(
                 text, voice=voice_path, speed=speed, model=self._model
diff --git a/api/src/services/audio.py b/api/src/services/audio.py
index 2055c9f..64062b8 100644
--- a/api/src/services/audio.py
+++ b/api/src/services/audio.py
@@ -4,10 +4,12 @@ import struct
 from io import BytesIO
 
 import numpy as np
+import math
 import scipy.io.wavfile as wavfile
 import soundfile as sf
 from loguru import logger
 from pydub import AudioSegment
+from torch import norm
 
 from ..core.config import settings
 from .streaming_audio_writer import StreamingAudioWriter
@@ -20,23 +22,66 @@ class AudioNormalizer:
         self.chunk_trim_ms = settings.gap_trim_ms
         self.sample_rate = 24000  # Sample rate of the audio
         self.samples_to_trim = int(self.chunk_trim_ms * self.sample_rate / 1000)
+        self.samples_to_pad_start= int(50 * self.sample_rate / 1000)
+
+    def find_first_last_non_silent(self,audio_data: np.ndarray, chunk_text: str, speed: float, silence_threshold_db: int = -45, is_last_chunk: bool = False) -> tuple[int, int]:
+        """Finds the indices of the first and last non-silent samples in audio data.
+        
+        Args:
+            audio_data: Input audio data as numpy array
+            chunk_text: The text sent to the model to generate the resulting speech
+            speed: The speaking speed of the voice
+            silence_threshold_db: How quiet audio has to be to be conssidered silent
+            is_last_chunk: Whether this is the last chunk
+            
+        Returns:
+            A tuple with the start of the non silent portion and with the end of the non silent portion
+        """
+
+        pad_multiplier=1
+        split_character=chunk_text.strip()
+        if len(split_character) > 0:
+            split_character=split_character[-1]
+            if split_character in settings.dynamic_gap_trim_padding_char_multiplier:
+                pad_multiplier=settings.dynamic_gap_trim_padding_char_multiplier[split_character]
+
+        if not is_last_chunk:
+            samples_to_pad_end= max(int((settings.dynamic_gap_trim_padding_ms * self.sample_rate * pad_multiplier) / 1000) - self.samples_to_pad_start, 0)
+        else:
+            samples_to_pad_end=self.samples_to_pad_start
+        # Convert dBFS threshold to amplitude
+        amplitude_threshold = np.iinfo(audio_data.dtype).max * (10 ** (silence_threshold_db / 20))
+        # Find the first samples above the silence threshold at the start and end of the audio
+        non_silent_index_start, non_silent_index_end = None,None 
+
+        for X in range(0,len(audio_data)):
+            #print(audio_data[X])
+            if audio_data[X] > amplitude_threshold:
+                non_silent_index_start=X
+                break
+        
+        for X in range(len(audio_data) - 1, -1, -1):
+            if audio_data[X] > amplitude_threshold:
+                non_silent_index_end=X
+                break
+
+        # Handle the case where the entire audio is silent
+        if non_silent_index_start == None or non_silent_index_end == None:
+            return 0, len(audio_data)
+
+        return max(non_silent_index_start - self.samples_to_pad_start,0), min(non_silent_index_end + math.ceil(samples_to_pad_end / speed),len(audio_data))
 
     async def normalize(self, audio_data: np.ndarray) -> np.ndarray:
-        """Convert audio data to int16 range and trim silence from start and end
+        """Convert audio data to int16 range
 
         Args:
             audio_data: Input audio data as numpy array
-
         Returns:
-            Normalized and trimmed audio data
+            Normalized audio data
         """
         if len(audio_data) == 0:
             raise ValueError("Empty audio data")
 
-        # Trim start and end if enough samples
-        if len(audio_data) > (2 * self.samples_to_trim):
-            audio_data = audio_data[self.samples_to_trim : -self.samples_to_trim]
-
         # Scale directly to int16 range with clipping
         return np.clip(audio_data * 32767, -32768, 32767).astype(np.int16)
 
@@ -71,6 +116,8 @@ class AudioService:
         audio_data: np.ndarray,
         sample_rate: int,
         output_format: str,
+        speed: float = 1,
+        chunk_text: str = "",
         is_first_chunk: bool = True,
         is_last_chunk: bool = False,
         normalizer: AudioNormalizer = None,
@@ -81,6 +128,8 @@ class AudioService:
             audio_data: Numpy array of audio samples
             sample_rate: Sample rate of the audio
             output_format: Target format (wav, mp3, ogg, pcm)
+            speed: The speaking speed of the voice
+            chunk_text: The text sent to the model to generate the resulting speech
             is_first_chunk: Whether this is the first chunk
             is_last_chunk: Whether this is the last chunk
             normalizer: Optional AudioNormalizer instance for consistent normalization
@@ -96,8 +145,10 @@ class AudioService:
             # Always normalize audio to ensure proper amplitude scaling
             if normalizer is None:
                 normalizer = AudioNormalizer()
+                
             normalized_audio = await normalizer.normalize(audio_data)
-
+            normalized_audio = AudioService.trim_audio(normalized_audio,chunk_text,speed,is_last_chunk,normalizer)
+            
             # Get or create format-specific writer
             writer_key = f"{output_format}_{sample_rate}"
             if is_first_chunk or writer_key not in AudioService._writers:
@@ -123,3 +174,27 @@ class AudioService:
             raise ValueError(
                 f"Failed to convert audio stream to {output_format}: {str(e)}"
             )
+    @staticmethod
+    def trim_audio(audio_data: np.ndarray, chunk_text: str = "", speed: float = 1, is_last_chunk: bool = False, normalizer: AudioNormalizer = None) -> np.ndarray:
+        """Trim silence from start and end
+
+        Args:
+            audio_data: Input audio data as numpy array
+            chunk_text: The text sent to the model to generate the resulting speech
+            speed: The speaking speed of the voice
+            is_last_chunk: Whether this is the last chunk
+            normalizer: Optional AudioNormalizer instance for consistent normalization
+            
+        Returns:
+            Trimmed audio data
+        """
+        if normalizer is None:
+            normalizer = AudioNormalizer()
+        
+        # Trim start and end if enough samples
+        if len(audio_data) > (2 * normalizer.samples_to_trim):
+            audio_data = audio_data[normalizer.samples_to_trim : -normalizer.samples_to_trim]
+            
+        # Find non silent portion and trim 
+        start_index,end_index=normalizer.find_first_last_non_silent(audio_data,chunk_text,speed,is_last_chunk=is_last_chunk)
+        return audio_data[start_index:end_index]
\ No newline at end of file
diff --git a/api/src/services/text_processing/normalizer.py b/api/src/services/text_processing/normalizer.py
index 383abbd..7c728fb 100644
--- a/api/src/services/text_processing/normalizer.py
+++ b/api/src/services/text_processing/normalizer.py
@@ -6,6 +6,7 @@ Converts them into a format suitable for text-to-speech processing.
 
 import re
 from functools import lru_cache
+import inflect
 
 # Constants
 VALID_TLDS = [
@@ -50,6 +51,26 @@ VALID_TLDS = [
     "io",
 ]
 
+VALID_UNITS = {
+    "m":"meter", "cm":"centimeter", "mm":"millimeter", "km":"kilometer", "in":"inch", "ft":"foot", "yd":"yard", "mi":"mile",  # Length
+    "g":"gram", "kg":"kilogram", "mg":"miligram",      # Mass
+    "s":"second", "ms":"milisecond", "min":"minutes", "h":"hour", # Time
+    "l":"liter", "ml":"mililiter", "cl":"centiliter", "dl":"deciliter",  # Volume
+    "kph":"kilometer per hour", "mph":"mile per hour","mi/h":"mile per hour", "m/s":"meter per second", "km/h":"kilometer per hour", "mm/s":"milimeter per second","cm/s":"centimeter per second", "ft/s":"feet per second", # Speed
+    "°c":"degree celsius","c":"degree celsius", "°f":"degree fahrenheit","f":"degree fahrenheit", "k":"kelvin",     # Temperature
+    "pa":"pascal", "kpa":"kilopascal", "mpa":"megapascal", "atm":"atmosphere",  # Pressure
+    "hz":"hertz", "khz":"kilohertz", "mhz":"megahertz", "ghz":"gigahertz", # Frequency
+    "v":"volt", "kv":"kilovolt", "mv":"mergavolt",      # Voltage
+    "a":"amp", "ma":"megaamp", "ka":"kiloamp",      # Current
+    "w":"watt", "kw":"kilowatt", "mw":"megawatt",      # Power
+    "j":"joule", "kj":"kilojoule", "mj":"megajoule",      # Energy
+    "Ω":"ohm", "kΩ":"kiloohm", "mΩ":"megaohm",      # Resistance (Ohm)
+    "f":"farad", "µf":"microfarad", "nf":"nanofarad", "pf":"picofarad", # Capacitance
+    "b":"byte", "kb":"kilobyte", "mb":"megabyte", "gb":"gigabyte", "tb":"terabyte", "pb":"petabyte", # Data size
+    "kbps":"kilobyte per second","mbps":"megabyte per second","gbps":"gigabyte per second",
+    "px":"pixel"  # CSS units
+}
+
 # Pre-compiled regex patterns for performance
 EMAIL_PATTERN = re.compile(
     r"\b[a-zA-Z0-9._%+-]+@[a-zA-Z0-9.-]+\.[a-z]{2,}\b", re.IGNORECASE
@@ -61,6 +82,9 @@ URL_PATTERN = re.compile(
     re.IGNORECASE,
 )
 
+UNIT_PATTERN = re.compile(r"((?<!\w)([+-]?)(\d{1,3}(,\d{3})*|\d+)(\.\d+)?)\s*(" + "|".join(sorted(list(VALID_UNITS.keys()),reverse=True)) + r"""){1}(?=[!"#$%&'()*+,-./:;<=>?@\[\\\]^_`{\|}~ \n]{1})""",re.IGNORECASE)
+
+INFLECT_ENGINE=inflect.engine()
 
 def split_num(num: re.Match[str]) -> str:
     """Handle number splitting for various formats"""
@@ -86,6 +110,13 @@ def split_num(num: re.Match[str]) -> str:
             return f"{left} oh {right}{s}"
     return f"{left} {right}{s}"
 
+def handle_units(u: re.Match[str]) -> str:
+    unit=u.group(6).strip() 
+    if unit.lower() in VALID_UNITS:
+        unit=VALID_UNITS[unit.lower()].split(" ")
+        number=u.group(1).strip()
+        unit[0]=INFLECT_ENGINE.no(unit[0],number)
+    return " ".join(unit)
 
 def handle_money(m: re.Match[str]) -> str:
     """Convert money expressions to spoken form"""
@@ -187,14 +218,17 @@ def normalize_text(text: str) -> str:
     # Pre-process URLs first
     text = normalize_urls(text)
 
+    # Pre-process numbers with units
+    text=UNIT_PATTERN.sub(handle_units,text)
+    
     # Replace quotes and brackets
     text = text.replace(chr(8216), "'").replace(chr(8217), "'")
     text = text.replace("«", chr(8220)).replace("»", chr(8221))
     text = text.replace(chr(8220), '"').replace(chr(8221), '"')
     text = text.replace("(", "«").replace(")", "»")
 
-    # Handle CJK punctuation
-    for a, b in zip("、。！，：；？", ",.!,:;?"):
+    # Handle CJK punctuation and some non standard chars
+    for a, b in zip("、。！，：；？–", ",.!,:;?-"):
         text = text.replace(a, b + " ")
 
     # Clean up whitespace
diff --git a/api/src/services/text_processing/text_processor.py b/api/src/services/text_processing/text_processor.py
index 924b014..0e3cfcf 100644
--- a/api/src/services/text_processing/text_processor.py
+++ b/api/src/services/text_processing/text_processor.py
@@ -26,7 +26,7 @@ def process_text_chunk(
         List of token IDs
     """
     start_time = time.time()
-
+    
     if skip_phonemize:
         # Input is already phonemes, just tokenize
         t0 = time.time()
@@ -35,12 +35,11 @@ def process_text_chunk(
     else:
         # Normal text processing pipeline
         t0 = time.time()
-        normalized = normalize_text(text)
         t1 = time.time()
 
         t0 = time.time()
         phonemes = phonemize(
-            normalized, language, normalize=False
+            text, language, normalize=False
         )  # Already normalized
         t1 = time.time()
 
@@ -50,7 +49,7 @@ def process_text_chunk(
 
     total_time = time.time() - start_time
     logger.debug(
-        f"Total processing took {total_time * 1000:.2f}ms for chunk: '{text[:50]}...'"
+        f"Total processing took {total_time * 1000:.2f}ms for chunk: '{text[:50]}{'...' if len(text) > 50 else ''}'"
     )
 
     return tokens
@@ -61,7 +60,7 @@ async def yield_chunk(
 ) -> Tuple[str, List[int]]:
     """Yield a chunk with consistent logging."""
     logger.debug(
-        f"Yielding chunk {chunk_count}: '{text[:50]}...' ({len(tokens)} tokens)"
+        f"Yielding chunk {chunk_count}: '{text[:50]}{'...' if len(text) > 50 else ''}' ({len(tokens)} tokens)"
     )
     return text, tokens
 
@@ -88,9 +87,10 @@ def process_text(text: str, language: str = "a") -> List[int]:
 
 def get_sentence_info(text: str) -> List[Tuple[str, List[int], int]]:
     """Process all sentences and return info."""
-    sentences = re.split(r"([.!?;:])", text)
+    if settings.advanced_text_normalization:
+        text=normalize_text(text)
+    sentences = re.split(r"([.!?;:])(?=\s|$)", text)
     results = []
-
     for i in range(0, len(sentences), 2):
         sentence = sentences[i].strip()
         punct = sentences[i + 1] if i + 1 < len(sentences) else ""
@@ -128,7 +128,7 @@ async def smart_split(
                 chunk_text = " ".join(current_chunk)
                 chunk_count += 1
                 logger.debug(
-                    f"Yielding chunk {chunk_count}: '{chunk_text[:50]}...' ({current_count} tokens)"
+                    f"Yielding chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({current_count} tokens)"
                 )
                 yield chunk_text, current_tokens
                 current_chunk = []
@@ -149,6 +149,7 @@ async def smart_split(
                     continue
 
                 full_clause = clause + comma
+                
                 tokens = process_text_chunk(full_clause)
                 count = len(tokens)
 
@@ -166,7 +167,7 @@ async def smart_split(
                         chunk_text = " ".join(clause_chunk)
                         chunk_count += 1
                         logger.debug(
-                            f"Yielding clause chunk {chunk_count}: '{chunk_text[:50]}...' ({clause_count} tokens)"
+                            f"Yielding clause chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({clause_count} tokens)"
                         )
                         yield chunk_text, clause_tokens
                     clause_chunk = [full_clause]
@@ -178,7 +179,7 @@ async def smart_split(
                 chunk_text = " ".join(clause_chunk)
                 chunk_count += 1
                 logger.debug(
-                    f"Yielding final clause chunk {chunk_count}: '{chunk_text[:50]}...' ({clause_count} tokens)"
+                    f"Yielding final clause chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({clause_count} tokens)"
                 )
                 yield chunk_text, clause_tokens
 
@@ -192,7 +193,7 @@ async def smart_split(
             chunk_text = " ".join(current_chunk)
             chunk_count += 1
             logger.info(
-                f"Yielding chunk {chunk_count}: '{chunk_text[:50]}...' ({current_count} tokens)"
+                f"Yielding chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({current_count} tokens)"
             )
             yield chunk_text, current_tokens
             current_chunk = [sentence]
@@ -217,7 +218,7 @@ async def smart_split(
                 chunk_text = " ".join(current_chunk)
                 chunk_count += 1
                 logger.info(
-                    f"Yielding chunk {chunk_count}: '{chunk_text[:50]}...' ({current_count} tokens)"
+                    f"Yielding chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({current_count} tokens)"
                 )
                 yield chunk_text, current_tokens
             current_chunk = [sentence]
@@ -229,7 +230,7 @@ async def smart_split(
         chunk_text = " ".join(current_chunk)
         chunk_count += 1
         logger.info(
-            f"Yielding final chunk {chunk_count}: '{chunk_text[:50]}...' ({current_count} tokens)"
+            f"Yielding final chunk {chunk_count}: '{chunk_text[:50]}{'...' if len(text) > 50 else ''}' ({current_count} tokens)"
         )
         yield chunk_text, current_tokens
 
diff --git a/api/src/services/tts_service.py b/api/src/services/tts_service.py
index 44fb709..3d533d9 100644
--- a/api/src/services/tts_service.py
+++ b/api/src/services/tts_service.py
@@ -67,6 +67,8 @@ class TTSService:
                         np.array([0], dtype=np.float32),  # Dummy data for type checking
                         24000,
                         output_format,
+                        speed,
+                        "",
                         is_first_chunk=False,
                         normalizer=normalizer,
                         is_last_chunk=True,
@@ -97,15 +99,22 @@ class TTSService:
                                     chunk_audio,
                                     24000,
                                     output_format,
+                                    speed,
+                                    chunk_text,
                                     is_first_chunk=is_first,
-                                    normalizer=normalizer,
                                     is_last_chunk=is_last,
+                                    normalizer=normalizer,
                                 )
                                 yield converted
                             except Exception as e:
                                 logger.error(f"Failed to convert audio: {str(e)}")
                         else:
-                            yield chunk_audio
+                            trimmed = await AudioService.trim_audio(chunk_audio,
+                                                                    chunk_text,
+                                                                    speed,
+                                                                    is_last,
+                                                                    normalizer)
+                            yield trimmed
                 else:
                     # For legacy backends, load voice tensor
                     voice_tensor = await self._voice_manager.load_voice(
@@ -130,6 +139,8 @@ class TTSService:
                                 chunk_audio,
                                 24000,
                                 output_format,
+                                speed,
+                                chunk_text,
                                 is_first_chunk=is_first,
                                 normalizer=normalizer,
                                 is_last_chunk=is_last,
@@ -138,7 +149,12 @@ class TTSService:
                         except Exception as e:
                             logger.error(f"Failed to convert audio: {str(e)}")
                     else:
-                        yield chunk_audio
+                        trimmed = await AudioService.trim_audio(chunk_audio,
+                                                                    chunk_text,
+                                                                    speed,
+                                                                    is_last,
+                                                                    normalizer)
+                        yield trimmed
             except Exception as e:
                 logger.error(f"Failed to process tokens: {str(e)}")
 
diff --git a/docker/cpu/Dockerfile b/docker/cpu/Dockerfile
index 5d6f99e..369d008 100644
--- a/docker/cpu/Dockerfile
+++ b/docker/cpu/Dockerfile
@@ -9,10 +9,10 @@ RUN apt-get update && apt-get install -y \
     curl \
     ffmpeg \
     g++ \
-    && apt-get clean \
-    && rm -rf /var/lib/apt/lists/* \
-    && mkdir -p /usr/share/espeak-ng-data \
-    && ln -s /usr/lib/*/espeak-ng-data/* /usr/share/espeak-ng-data/
+&& apt-get clean \
+&& rm -rf /var/lib/apt/lists/* \
+&& mkdir -p /usr/share/espeak-ng-data \
+&& ln -s /usr/lib/*/espeak-ng-data/* /usr/share/espeak-ng-data/
 
 # Install UV using the installer script
 RUN curl -LsSf https://astral.sh/uv/install.sh | sh && \
@@ -20,7 +20,7 @@ RUN curl -LsSf https://astral.sh/uv/install.sh | sh && \
     mv /root/.local/bin/uvx /usr/local/bin/
 
 # Create non-root user and set up directories and permissions
-RUN useradd -m -u 1000 appuser && \
+RUN useradd -m -u 1001 appuser && \
     mkdir -p /app/api/src/models/v1_0 && \
     chown -R appuser:appuser /app
 
@@ -32,7 +32,7 @@ COPY --chown=appuser:appuser pyproject.toml ./pyproject.toml
 
 # Install dependencies
 RUN --mount=type=cache,target=/root/.cache/uv \
-    uv venv && \
+    uv venv --python 3.11 && \
     uv sync --extra cpu
 
 # Copy project files including models
@@ -40,6 +40,7 @@ COPY --chown=appuser:appuser api ./api
 COPY --chown=appuser:appuser web ./web
 COPY --chown=appuser:appuser docker/scripts/ ./
 RUN chmod +x ./entrypoint.sh
+RUN sed -i 's/\r$//' ./entrypoint.sh
 
 # Set environment variables
 ENV PYTHONUNBUFFERED=1 \
diff --git a/docker/gpu/Dockerfile b/docker/gpu/Dockerfile
index ce0f646..b19cf80 100644
--- a/docker/gpu/Dockerfile
+++ b/docker/gpu/Dockerfile
@@ -1,26 +1,29 @@
-FROM --platform=$BUILDPLATFORM nvidia/cuda:12.4.1-cudnn-runtime-ubuntu22.04
+FROM --platform=$BUILDPLATFORM nvidia/cuda:12.8.0-cudnn-runtime-ubuntu24.04
 # Set non-interactive frontend
 ENV DEBIAN_FRONTEND=noninteractive
 
 # Install Python and other dependencies
 RUN apt-get update && apt-get install -y \
     python3.10 \
-    python3.10-venv \
+    python3-venv \
     espeak-ng \
     espeak-ng-data \
     git \
     libsndfile1 \
     curl \
     ffmpeg \
- && apt-get clean && rm -rf /var/lib/apt/lists/* \
+ && apt-get clean \
+ && rm -rf /var/lib/apt/lists/* \
  && mkdir -p /usr/share/espeak-ng-data \
  && ln -s /usr/lib/*/espeak-ng-data/* /usr/share/espeak-ng-data/
 
 # Install UV using the installer script
 RUN curl -LsSf https://astral.sh/uv/install.sh | sh && \
     mv /root/.local/bin/uv /usr/local/bin/ && \
-    mv /root/.local/bin/uvx /usr/local/bin/ && \
-    useradd -m -u 1000 appuser && \
+    mv /root/.local/bin/uvx /usr/local/bin/
+    
+# Create non-root user and set up directories and permissions
+RUN useradd -m -u 1001 appuser && \
     mkdir -p /app/api/src/models/v1_0 && \
     chown -R appuser:appuser /app 
 
@@ -32,7 +35,7 @@ COPY --chown=appuser:appuser pyproject.toml ./pyproject.toml
 
 # Install dependencies with GPU extras (using cache mounts)
 RUN --mount=type=cache,target=/root/.cache/uv \
-    uv venv && \
+    uv venv --python 3.11 && \
     uv sync --extra gpu
 
 # Copy project files including models and sync again
@@ -40,6 +43,7 @@ COPY --chown=appuser:appuser api ./api
 COPY --chown=appuser:appuser web ./web
 COPY --chown=appuser:appuser docker/scripts/ ./
 RUN chmod +x ./entrypoint.sh
+RUN sed -i 's/\r$//' ./entrypoint.sh
 RUN --mount=type=cache,target=/root/.cache/uv \
     uv sync --extra gpu
 
diff --git a/pyproject.toml b/pyproject.toml
index 336519e..4205fea 100644
--- a/pyproject.toml
+++ b/pyproject.toml
@@ -36,7 +36,8 @@ dependencies = [
     "kokoro @ git+https://github.com/hexgrad/kokoro.git@31a2b6337b8c1b1418ef68c48142328f640da938",
     'misaki[en,ja,ko,zh] @ git+https://github.com/hexgrad/misaki.git@ebc76c21b66c5fc4866ed0ec234047177b396170',
     "spacy==3.7.2",
-    "en-core-web-sm @ https://github.com/explosion/spacy-models/releases/download/en_core_web_sm-3.7.1/en_core_web_sm-3.7.1-py3-none-any.whl"
+    "en-core-web-sm @ https://github.com/explosion/spacy-models/releases/download/en_core_web_sm-3.7.1/en_core_web_sm-3.7.1-py3-none-any.whl",
+    "inflect>=7.5.0",
 ]
 
 [project.optional-dependencies]
diff --git a/start-gpu.bat b/start-gpu.bat
new file mode 100644
index 0000000..d7067e4
--- /dev/null
+++ b/start-gpu.bat
@@ -0,0 +1,10 @@
+set PYTHONUTF8=1
+set USE_GPU=true
+set USE_ONNX=false
+set PYTHONPATH=%PROJECT_ROOT%;%PROJECT_ROOT%\api
+set MODEL_DIR=src\models
+set VOICES_DIR=src\voices\v1_0
+set WEB_PLAYER_PATH=%PROJECT_ROOT%\web
+
+call uv pip install -e ".[gpu]"
+call uv run uvicorn api.src.main:app --reload --host 0.0.0.0 --port 8880
\ No newline at end of file