import os
import sys
import requests
import json
import time
import wave
import numpy as np
import sounddevice as sd
import argparse
import threading
import queue
import asyncio
from concurrent.futures import ThreadPoolExecutor
from typing import List, Dict, Any, Optional, Generator, Union, Tuple
from dotenv import load_dotenv

# Helper to detect if running in Uvicorn's reloader
def is_reloader_process():
    """Check if the current process is a uvicorn reloader"""
    return (sys.argv[0].endswith('_continuation.py') or 
            os.environ.get('UVICORN_STARTED') == 'true')

# Set a flag to avoid repeat messages
IS_RELOADER = is_reloader_process()
if not IS_RELOADER:
    os.environ['UVICORN_STARTED'] = 'true'

# Load environment variables from .env file
load_dotenv()

# Detect hardware capabilities and display information
import torch
import psutil

# Detect if we're on a high-end system based on hardware capabilities
HIGH_END_GPU = False
if torch.cuda.is_available():
    # Get GPU properties
    props = torch.cuda.get_device_properties(0)
    gpu_name = props.name
    gpu_mem_gb = props.total_memory / (1024**3)
    compute_capability = f"{props.major}.{props.minor}"
    
    # Consider high-end if: large VRAM (≥16GB) OR high compute capability (≥8.0) OR large VRAM (≥12GB) with good CC (≥7.0)
    HIGH_END_GPU = (gpu_mem_gb >= 16.0 or 
                    props.major >= 8 or 
                    (gpu_mem_gb >= 12.0 and props.major >= 7))
        
    if HIGH_END_GPU:
        if not IS_RELOADER:
            print(f"🖥️ Hardware: High-end CUDA GPU detected")
            print(f"📊 Device: {gpu_name}")
            print(f"📊 VRAM: {gpu_mem_gb:.2f} GB")
            print(f"📊 Compute Capability: {compute_capability}")
            print("🚀 Using high-performance optimizations")
    else:
        if not IS_RELOADER:
            print(f"🖥️ Hardware: CUDA GPU detected")
            print(f"📊 Device: {gpu_name}")
            print(f"📊 VRAM: {gpu_mem_gb:.2f} GB")
            print(f"📊 Compute Capability: {compute_capability}")
            print("🚀 Using GPU-optimized settings")
else:
    # Get CPU info
    cpu_cores = psutil.cpu_count(logical=False)
    cpu_threads = psutil.cpu_count(logical=True)
    ram_gb = psutil.virtual_memory().total / (1024**3)
    
    if not IS_RELOADER:
        print(f"🖥️ Hardware: CPU only (No CUDA GPU detected)")
        print(f"📊 CPU: {cpu_cores} cores, {cpu_threads} threads")
        print(f"📊 RAM: {ram_gb:.2f} GB")
        print("⚙️ Using CPU-optimized settings")

# Load configuration from environment variables without hardcoded defaults
# Critical settings - will log errors if missing
required_settings = ["ORPHEUS_API_URL"]
missing_settings = [s for s in required_settings if s not in os.environ]
if missing_settings:
    print(f"ERROR: Missing required environment variable(s): {', '.join(missing_settings)}")
    print("Please set them in .env file or environment. See .env.example for defaults.")

# API connection settings
API_URL = os.environ.get("ORPHEUS_API_URL")
if not API_URL:
    print("WARNING: ORPHEUS_API_URL not set. API calls will fail until configured.")

HEADERS = {
    "Content-Type": "application/json"
}

# Request timeout settings
try:
    REQUEST_TIMEOUT = int(os.environ.get("ORPHEUS_API_TIMEOUT", "120"))
except (ValueError, TypeError):
    print("WARNING: Invalid ORPHEUS_API_TIMEOUT value, using 120 seconds as fallback")
    REQUEST_TIMEOUT = 120

# Model generation parameters from environment variables
try:
    MAX_TOKENS = int(os.environ.get("ORPHEUS_MAX_TOKENS", "8192"))
except (ValueError, TypeError):
    print("WARNING: Invalid ORPHEUS_MAX_TOKENS value, using 8192 as fallback")
    MAX_TOKENS = 8192

try:
    TEMPERATURE = float(os.environ.get("ORPHEUS_TEMPERATURE", "0.6"))
except (ValueError, TypeError):
    print("WARNING: Invalid ORPHEUS_TEMPERATURE value, using 0.6 as fallback")
    TEMPERATURE = 0.6

try:
    TOP_P = float(os.environ.get("ORPHEUS_TOP_P", "0.9"))
except (ValueError, TypeError):
    print("WARNING: Invalid ORPHEUS_TOP_P value, using 0.9 as fallback")
    TOP_P = 0.9

# Repetition penalty is hardcoded to 1.1 which is the only stable value for quality output
REPETITION_PENALTY = 1.1

try:
    SAMPLE_RATE = int(os.environ.get("ORPHEUS_SAMPLE_RATE", "24000"))
except (ValueError, TypeError):
    print("WARNING: Invalid ORPHEUS_SAMPLE_RATE value, using 24000 as fallback")
    SAMPLE_RATE = 24000

# Print loaded configuration only in the main process, not in the reloader
if not IS_RELOADER:
    print(f"Configuration loaded:")
    print(f"  API_URL: {API_URL}")
    print(f"  MAX_TOKENS: {MAX_TOKENS}")
    print(f"  TEMPERATURE: {TEMPERATURE}")
    print(f"  TOP_P: {TOP_P}")
    print(f"  REPETITION_PENALTY: {REPETITION_PENALTY}")

# Parallel processing settings
NUM_WORKERS = 4 if HIGH_END_GPU else 2

# Define voices by language
ENGLISH_VOICES = ["tara", "leah", "jess", "leo", "dan", "mia", "zac", "zoe"]
FRENCH_VOICES = ["pierre", "amelie", "marie"]
GERMAN_VOICES = ["jana", "thomas", "max"]
KOREAN_VOICES = ["유나", "준서"]
HINDI_VOICES = ["ऋतिका"]
MANDARIN_VOICES = ["长乐", "白芷"]
SPANISH_VOICES = ["javi", "sergio", "maria"]
ITALIAN_VOICES = ["pietro", "giulia", "carlo"]

# Combined list for API compatibility
AVAILABLE_VOICES = (
    ENGLISH_VOICES + 
    FRENCH_VOICES + 
    GERMAN_VOICES + 
    KOREAN_VOICES + 
    HINDI_VOICES + 
    MANDARIN_VOICES + 
    SPANISH_VOICES + 
    ITALIAN_VOICES
)
DEFAULT_VOICE = "tara"  # Best voice according to documentation

# Map voices to languages for the UI
VOICE_TO_LANGUAGE = {}
VOICE_TO_LANGUAGE.update({voice: "english" for voice in ENGLISH_VOICES})
VOICE_TO_LANGUAGE.update({voice: "french" for voice in FRENCH_VOICES})
VOICE_TO_LANGUAGE.update({voice: "german" for voice in GERMAN_VOICES})
VOICE_TO_LANGUAGE.update({voice: "korean" for voice in KOREAN_VOICES})
VOICE_TO_LANGUAGE.update({voice: "hindi" for voice in HINDI_VOICES})
VOICE_TO_LANGUAGE.update({voice: "mandarin" for voice in MANDARIN_VOICES})
VOICE_TO_LANGUAGE.update({voice: "spanish" for voice in SPANISH_VOICES})
VOICE_TO_LANGUAGE.update({voice: "italian" for voice in ITALIAN_VOICES})

# Languages list for the UI
AVAILABLE_LANGUAGES = ["english", "french", "german", "korean", "hindi", "mandarin", "spanish", "italian"]

# Import the unified token handling from speechpipe
from .speechpipe import turn_token_into_id, CUSTOM_TOKEN_PREFIX

# Special token IDs for Orpheus model
START_TOKEN_ID = 128259
END_TOKEN_IDS = [128009, 128260, 128261, 128257]

# Performance monitoring
class PerformanceMonitor:
    """Track and report performance metrics"""
    def __init__(self):
        self.start_time = time.time()
        self.token_count = 0
        self.audio_chunks = 0
        self.last_report_time = time.time()
        self.report_interval = 2.0  # seconds
        
    def add_tokens(self, count: int = 1) -> None:
        self.token_count += count
        self._check_report()
        
    def add_audio_chunk(self) -> None:
        self.audio_chunks += 1
        self._check_report()
        
    def _check_report(self) -> None:
        current_time = time.time()
        if current_time - self.last_report_time >= self.report_interval:
            self.report()
            self.last_report_time = current_time
            
    def report(self) -> None:
        elapsed = time.time() - self.start_time
        if elapsed < 0.001:
            return
            
        tokens_per_sec = self.token_count / elapsed
        chunks_per_sec = self.audio_chunks / elapsed
        
        # Estimate audio duration based on audio chunks (each chunk is ~0.085s of audio)
        est_duration = self.audio_chunks * 0.085
        
        print(f"Progress: {tokens_per_sec:.1f} tokens/sec, est. {est_duration:.1f}s audio generated, {self.token_count} tokens, {self.audio_chunks} chunks in {elapsed:.1f}s")

# Create global performance monitor
perf_monitor = PerformanceMonitor()

def format_prompt(prompt: str, voice: str = DEFAULT_VOICE) -> str:
    """Format prompt for Orpheus model with voice prefix and special tokens."""
    # Validate voice and provide fallback
    if voice not in AVAILABLE_VOICES:
        print(f"Warning: Voice '{voice}' not recognized. Using '{DEFAULT_VOICE}' instead.")
        voice = DEFAULT_VOICE
        
    # Format similar to how engine_class.py does it with special tokens
    formatted_prompt = f"{voice}: {prompt}"
    
    # Add special token markers for the Orpheus-FASTAPI
    special_start = "<|audio|>"  # Using the additional_special_token from config
    special_end = "<|eot_id|>"   # Using the eos_token from config
    
    return f"{special_start}{formatted_prompt}{special_end}"

def generate_tokens_from_api(prompt: str, voice: str = DEFAULT_VOICE, temperature: float = TEMPERATURE, 
                           top_p: float = TOP_P, max_tokens: int = MAX_TOKENS, 
                           repetition_penalty: float = REPETITION_PENALTY) -> Generator[str, None, None]:
    """Generate tokens from text using OpenAI-compatible API with optimized streaming and retry logic."""
    start_time = time.time()
    formatted_prompt = format_prompt(prompt, voice)
    print(f"Generating speech for: {formatted_prompt}")
    
    # Optimize the token generation for GPUs
    if HIGH_END_GPU:
        # Use more aggressive parameters for faster generation on high-end GPUs
        print("Using optimized parameters for high-end GPU")
    elif torch.cuda.is_available():
        print("Using optimized parameters for GPU acceleration")
    
    # Create the request payload (model field may not be required by some endpoints but included for compatibility)
    payload = {
        "prompt": formatted_prompt,
        "max_tokens": max_tokens,
        "temperature": temperature,
        "top_p": top_p,
        "repeat_penalty": repetition_penalty,
        "stream": True  # Always stream for better performance
    }
    
    # Add model field - this is ignored by many local inference servers for /v1/completions
    # but included for compatibility with OpenAI API and some servers that may use it
    model_name = os.environ.get("ORPHEUS_MODEL_NAME", "Orpheus-3b-FT-Q8_0.gguf")
    payload["model"] = model_name
    
    # Session for connection pooling and retry logic
    session = requests.Session()
    
    retry_count = 0
    max_retries = 3
    
    while retry_count < max_retries:
        try:
            # Make the API request with streaming and timeout
            response = session.post(
                API_URL, 
                headers=HEADERS, 
                json=payload, 
                stream=True,
                timeout=REQUEST_TIMEOUT
            )
            
            if response.status_code != 200:
                print(f"Error: API request failed with status code {response.status_code}")
                print(f"Error details: {response.text}")
                # Retry on server errors (5xx) but not on client errors (4xx)
                if response.status_code >= 500:
                    retry_count += 1
                    wait_time = 2 ** retry_count  # Exponential backoff
                    print(f"Retrying in {wait_time} seconds...")
                    time.sleep(wait_time)
                    continue
                return
            
            # Process the streamed response with better buffering
            buffer = ""
            token_counter = 0
            
            # Iterate through the response to get tokens
            for line in response.iter_lines():
                if line:
                    line_str = line.decode('utf-8')
                    if line_str.startswith('data: '):
                        data_str = line_str[6:]  # Remove the 'data: ' prefix
                        
                        if data_str.strip() == '[DONE]':
                            break
                            
                        try:
                            data = json.loads(data_str)
                            if 'choices' in data and len(data['choices']) > 0:
                                token_chunk = data['choices'][0].get('text', '')
                                for token_text in token_chunk.split('>'):
                                    token_text = f'{token_text}>'
                                    token_counter += 1
                                    perf_monitor.add_tokens()

                                    if token_text:
                                        yield token_text
                        except json.JSONDecodeError as e:
                            print(f"Error decoding JSON: {e}")
                            continue
            
            # Generation completed successfully
            generation_time = time.time() - start_time
            tokens_per_second = token_counter / generation_time if generation_time > 0 else 0
            print(f"Token generation complete: {token_counter} tokens in {generation_time:.2f}s ({tokens_per_second:.1f} tokens/sec)")
            return
            
        except requests.exceptions.Timeout:
            print(f"Request timed out after {REQUEST_TIMEOUT} seconds")
            retry_count += 1
            if retry_count < max_retries:
                wait_time = 2 ** retry_count
                print(f"Retrying in {wait_time} seconds... (attempt {retry_count+1}/{max_retries})")
                time.sleep(wait_time)
            else:
                print("Max retries reached. Token generation failed.")
                return
                
        except requests.exceptions.ConnectionError:
            print(f"Connection error to API at {API_URL}")
            retry_count += 1
            if retry_count < max_retries:
                wait_time = 2 ** retry_count
                print(f"Retrying in {wait_time} seconds... (attempt {retry_count+1}/{max_retries})")
                time.sleep(wait_time)
            else:
                print("Max retries reached. Token generation failed.")
                return

# The turn_token_into_id function is now imported from speechpipe.py
# This eliminates duplicate code and ensures consistent behavior

def convert_to_audio(multiframe: List[int], count: int) -> Optional[bytes]:
    """Convert token frames to audio with performance monitoring."""
    # Import here to avoid circular imports
    from .speechpipe import convert_to_audio as orpheus_convert_to_audio
    start_time = time.time()
    result = orpheus_convert_to_audio(multiframe, count)
    
    if result is not None:
        perf_monitor.add_audio_chunk()
        
    return result

async def tokens_decoder(token_gen) -> Generator[bytes, None, None]:
    """Simplified token decoder with early first-chunk processing for lower latency."""
    buffer = []
    count = 0
    
    # Use different thresholds for first chunk vs. subsequent chunks
    first_chunk_processed = False
    min_frames_first = 7  # Process after just 7 tokens for first chunk (ultra-low latency)
    min_frames_subsequent = 28  # Default for reliability after first chunk (4 chunks of 7)
    process_every = 7  # Process every 7 tokens (standard for Orpheus model)
    
    start_time = time.time()
    last_log_time = start_time
    token_count = 0
    
    async for token_text in token_gen:
        token = turn_token_into_id(token_text, count)
        if token is not None and token > 0:
            # Add to buffer using simple append (reliable method)
            buffer.append(token)
            count += 1
            token_count += 1
            
            # Log throughput periodically
            current_time = time.time()
            if current_time - last_log_time > 5.0:  # Every 5 seconds
                elapsed = current_time - start_time
                if elapsed > 0:
                    print(f"Token processing rate: {token_count/elapsed:.1f} tokens/second")
                last_log_time = current_time
            
            # Different processing paths based on whether first chunk has been processed
            if not first_chunk_processed:
                # For first audio output, process as soon as we have enough tokens for one chunk
                if count >= min_frames_first:
                    buffer_to_proc = buffer[-min_frames_first:]
                    
                    # Process the first chunk for immediate audio feedback
                    print(f"Processing first audio chunk with {len(buffer_to_proc)} tokens")
                    audio_samples = convert_to_audio(buffer_to_proc, count)
                    if audio_samples is not None:
                        first_chunk_processed = True  # Mark first chunk as processed
                        yield audio_samples
            else:
                # For subsequent chunks, use standard processing with larger batch
                if count % process_every == 0 and count >= min_frames_subsequent:
                    # Use simple slice operation - reliable and correct
                    buffer_to_proc = buffer[-min_frames_subsequent:]
                    
                    # Debug output to help diagnose issues
                    if count % 28 == 0:
                        print(f"Processing buffer with {len(buffer_to_proc)} tokens, total collected: {len(buffer)}")
                    
                    # Process the tokens
                    audio_samples = convert_to_audio(buffer_to_proc, count)
                    if audio_samples is not None:
                        yield audio_samples

def tokens_decoder_sync(syn_token_gen, output_file=None):
    """Optimized synchronous wrapper with parallel processing and efficient file I/O."""
    # Use a larger queue for high-end systems
    queue_size = 100 if HIGH_END_GPU else 50
    audio_queue = queue.Queue(maxsize=queue_size)
    audio_segments = []
    
    # If output_file is provided, prepare WAV file with buffered I/O
    wav_file = None
    if output_file:
        # Create directory if it doesn't exist
        os.makedirs(os.path.dirname(os.path.abspath(output_file)), exist_ok=True)
        wav_file = wave.open(output_file, "wb")
        wav_file.setnchannels(1)
        wav_file.setsampwidth(2)
        wav_file.setframerate(SAMPLE_RATE)
    
    # Batch processing of tokens for improved throughput
    batch_size = 32 if HIGH_END_GPU else 16
    
    # Thread synchronization for proper completion detection
    producer_done_event = threading.Event()
    producer_started_event = threading.Event()
    
    # Convert the synchronous token generator into an async generator with batching
    async def async_token_gen():
        batch = []
        for token in syn_token_gen:
            batch.append(token)
            if len(batch) >= batch_size:
                for t in batch:
                    yield t
                batch = []
        # Process any remaining tokens in the final batch
        for t in batch:
            yield t

    async def async_producer():
        # Track performance with more granular metrics
        start_time = time.time()
        chunk_count = 0
        last_log_time = start_time
        
        try:
            # Signal that producer has started processing
            producer_started_event.set()
            
            async for audio_chunk in tokens_decoder(async_token_gen()):
                # Process each audio chunk from the decoder
                if audio_chunk:
                    audio_queue.put(audio_chunk)
                    chunk_count += 1
                    
                    # Log performance periodically
                    current_time = time.time()
                    if current_time - last_log_time >= 3.0:  # Every 3 seconds
                        elapsed = current_time - last_log_time
                        if elapsed > 0:
                            recent_chunks = chunk_count
                            chunks_per_sec = recent_chunks / elapsed
                            print(f"Audio generation rate: {chunks_per_sec:.2f} chunks/second")
                        last_log_time = current_time
                        # Reset chunk counter for next interval
                        chunk_count = 0
        except Exception as e:
            print(f"Error in token processing: {str(e)}")
            import traceback
            traceback.print_exc()
        finally:
            # Always signal completion, even if there was an error
            print("Producer completed - setting done event")
            producer_done_event.set()
            # Add sentinel to queue to signal end of stream
            audio_queue.put(None)

    def run_async():
        """Run the async producer in its own thread"""
        asyncio.run(async_producer())

    # Use a separate thread with higher priority for producer
    thread = threading.Thread(target=run_async, name="TokenProcessor")
    thread.daemon = True  # Allow thread to be terminated when main thread exits
    thread.start()
    
    # Wait for producer to actually start before proceeding
    # This avoids race conditions where we might try to read from an empty queue
    # before the producer has had a chance to add anything
    producer_started_event.wait(timeout=5.0)
    
    # Optimized I/O approach for all systems
    # This approach is simpler and more reliable than separate code paths
    write_buffer = bytearray()
    buffer_max_size = 1024 * 1024  # 1MB max buffer size (adjustable)
    
    # Keep track of the last time we checked for completion
    last_check_time = time.time()
    check_interval = 1.0  # Check producer status every second
    
    # Process audio chunks until we're done
    while True:
        try:
            # Get the next audio chunk with a short timeout
            # This allows us to periodically check status and handle other events
            audio = audio_queue.get(timeout=0.1)
            
            # None marker indicates end of stream
            if audio is None:
                print("Received end-of-stream marker")
                break
            
            # Store the audio segment for return value
            audio_segments.append(audio)
            
            # Write to file if needed
            if wav_file:
                write_buffer.extend(audio)
                
                # Flush buffer if it's large enough
                if len(write_buffer) >= buffer_max_size:
                    wav_file.writeframes(write_buffer)
                    write_buffer = bytearray()  # Reset buffer
        
        except queue.Empty:
            # No data available right now
            current_time = time.time()
            
            # Periodically check if producer is done
            if current_time - last_check_time > check_interval:
                last_check_time = current_time
                
                # If producer is done and queue is empty, we're finished
                if producer_done_event.is_set() and audio_queue.empty():
                    print("Producer done and queue empty - finishing consumer")
                    break
                
                # Flush buffer periodically even if not full
                if wav_file and len(write_buffer) > 0:
                    wav_file.writeframes(write_buffer)
                    write_buffer = bytearray()  # Reset buffer
    
    # Extra safety check - ensure thread is done
    if thread.is_alive():
        print("Waiting for token processor thread to complete...")
        thread.join(timeout=10.0)
        if thread.is_alive():
            print("WARNING: Token processor thread did not complete within timeout")
    
    # Final flush of any remaining data
    if wav_file and len(write_buffer) > 0:
        print(f"Final buffer flush: {len(write_buffer)} bytes")
        wav_file.writeframes(write_buffer)
    
    # Close WAV file if opened
    if wav_file:
        wav_file.close()
        if output_file:
            print(f"Audio saved to {output_file}")
    
    # Calculate and print detailed performance metrics
    if audio_segments:
        total_bytes = sum(len(segment) for segment in audio_segments)
        duration = total_bytes / (2 * SAMPLE_RATE)  # 2 bytes per sample at 24kHz
        total_time = time.time() - perf_monitor.start_time
        realtime_factor = duration / total_time if total_time > 0 else 0
        
        print(f"Generated {len(audio_segments)} audio segments")
        print(f"Generated {duration:.2f} seconds of audio in {total_time:.2f} seconds")
        print(f"Realtime factor: {realtime_factor:.2f}x")
        
        if realtime_factor < 1.0:
            print("⚠️ Warning: Generation is slower than realtime")
        else:
            print(f"✓ Generation is {realtime_factor:.1f}x faster than realtime")
    
    return audio_segments

def stream_audio(audio_buffer):
    """Stream audio buffer to output device with error handling."""
    if audio_buffer is None or len(audio_buffer) == 0:
        return
    
    try:
        # Convert bytes to NumPy array (16-bit PCM)
        audio_data = np.frombuffer(audio_buffer, dtype=np.int16)
        
        # Normalize to float in range [-1, 1] for playback
        audio_float = audio_data.astype(np.float32) / 32767.0
        
        # Play the audio with proper device selection and error handling
        sd.play(audio_float, SAMPLE_RATE)
        sd.wait()
    except Exception as e:
        print(f"Audio playback error: {e}")

import re
import numpy as np
from io import BytesIO
import wave

def split_text_into_sentences(text):
    """Split text into sentences with a more reliable approach."""
    # We'll use a simple approach that doesn't rely on variable-width lookbehinds
    # which aren't supported in Python's regex engine
    
    # First, split on common sentence ending punctuation
    # This isn't perfect but works for most cases and avoids the regex error
    parts = []
    current_sentence = ""
    
    for char in text:
        current_sentence += char
        
        # If we hit a sentence ending followed by a space, consider this a potential sentence end
        if char in (' ', '\n', '\t') and len(current_sentence) > 1:
            prev_char = current_sentence[-2]
            if prev_char in ('.', '!', '?'):
                # Check if this is likely a real sentence end and not an abbreviation
                # (Simple heuristic: if there's a space before the period, it's likely a real sentence end)
                if len(current_sentence) > 3 and current_sentence[-3] not in ('.', ' '):
                    parts.append(current_sentence.strip())
                    current_sentence = ""
    
    # Add any remaining text
    if current_sentence.strip():
        parts.append(current_sentence.strip())
    
    # Combine very short segments to avoid tiny audio files
    min_chars = 20  # Minimum reasonable sentence length
    combined_sentences = []
    i = 0
    
    while i < len(parts):
        current = parts[i]
        
        # If this is a short sentence and not the last one, combine with next
        while i < len(parts) - 1 and len(current) < min_chars:
            i += 1
            current += " " + parts[i]
            
        combined_sentences.append(current)
        i += 1
    
    return combined_sentences

def generate_speech_from_api(prompt, voice=DEFAULT_VOICE, output_file=None, temperature=TEMPERATURE, 
                     top_p=TOP_P, max_tokens=MAX_TOKENS, repetition_penalty=None, 
                     use_batching=True, max_batch_chars=1000):
    """Generate speech from text using Orpheus model with performance optimizations."""
    print(f"Starting speech generation for '{prompt[:50]}{'...' if len(prompt) > 50 else ''}'")
    print(f"Using voice: {voice}, GPU acceleration: {'Yes (High-end)' if HIGH_END_GPU else 'Yes' if torch.cuda.is_available() else 'No'}")
    
    # Reset performance monitor
    global perf_monitor
    perf_monitor = PerformanceMonitor()
    
    start_time = time.time()
    
    # For shorter text, use the standard non-batched approach
    if not use_batching or len(prompt) < max_batch_chars:
        # Note: we ignore any provided repetition_penalty and always use the hardcoded value
        # This ensures consistent quality regardless of what might be passed in
        result = tokens_decoder_sync(
            generate_tokens_from_api(
                prompt=prompt, 
                voice=voice,
                temperature=temperature,
                top_p=top_p,
                max_tokens=max_tokens,
                repetition_penalty=REPETITION_PENALTY  # Always use hardcoded value
            ),
            output_file=output_file
        )
        
        # Report final performance metrics
        end_time = time.time()
        total_time = end_time - start_time
        print(f"Total speech generation completed in {total_time:.2f} seconds")
        
        return result
    
    # For longer text, use sentence-based batching
    print(f"Using sentence-based batching for text with {len(prompt)} characters")
    
    # Split the text into sentences
    sentences = split_text_into_sentences(prompt)
    print(f"Split text into {len(sentences)} segments")
    
    # Create batches by combining sentences up to max_batch_chars
    batches = []
    current_batch = ""
    
    for sentence in sentences:
        # If adding this sentence would exceed the batch size, start a new batch
        if len(current_batch) + len(sentence) > max_batch_chars and current_batch:
            batches.append(current_batch)
            current_batch = sentence
        else:
            # Add separator space if needed
            if current_batch:
                current_batch += " "
            current_batch += sentence
    
    # Add the last batch if it's not empty
    if current_batch:
        batches.append(current_batch)
    
    print(f"Created {len(batches)} batches for processing")
    
    # Process each batch and collect audio segments
    all_audio_segments = []
    batch_temp_files = []
    
    for i, batch in enumerate(batches):
        print(f"Processing batch {i+1}/{len(batches)} ({len(batch)} characters)")
        
        # Create a temporary file for this batch if an output file is requested
        temp_output_file = None
        if output_file:
            temp_output_file = f"outputs/temp_batch_{i}_{int(time.time())}.wav"
            batch_temp_files.append(temp_output_file)
        
        # Generate speech for this batch
        batch_segments = tokens_decoder_sync(
            generate_tokens_from_api(
                prompt=batch,
                voice=voice,
                temperature=temperature,
                top_p=top_p,
                max_tokens=max_tokens,
                repetition_penalty=REPETITION_PENALTY
            ),
            output_file=temp_output_file
        )
        
        # Add to our collection
        all_audio_segments.extend(batch_segments)
    
    # If an output file was requested, stitch together the temporary files
    if output_file and batch_temp_files:
        # Stitch together WAV files
        stitch_wav_files(batch_temp_files, output_file)
        
        # Clean up temporary files
        for temp_file in batch_temp_files:
            try:
                os.remove(temp_file)
            except Exception as e:
                print(f"Warning: Could not remove temporary file {temp_file}: {e}")
    
    # Report final performance metrics
    end_time = time.time()
    total_time = end_time - start_time
    
    # Calculate combined duration
    if all_audio_segments:
        total_bytes = sum(len(segment) for segment in all_audio_segments)
        duration = total_bytes / (2 * SAMPLE_RATE)  # 2 bytes per sample at 24kHz
        print(f"Generated {len(all_audio_segments)} audio segments")
        print(f"Generated {duration:.2f} seconds of audio in {total_time:.2f} seconds")
        print(f"Realtime factor: {duration/total_time:.2f}x")
        
    print(f"Total speech generation completed in {total_time:.2f} seconds")
    
    return all_audio_segments

def stitch_wav_files(input_files, output_file, crossfade_ms=50):
    """Stitch multiple WAV files together with crossfading for smooth transitions."""
    if not input_files:
        return
        
    print(f"Stitching {len(input_files)} WAV files together with {crossfade_ms}ms crossfade")
    
    # If only one file, just copy it
    if len(input_files) == 1:
        import shutil
        shutil.copy(input_files[0], output_file)
        return
    
    # Convert crossfade_ms to samples
    crossfade_samples = int(SAMPLE_RATE * crossfade_ms / 1000)
    print(f"Using {crossfade_samples} samples for crossfade at {SAMPLE_RATE}Hz")
    
    # Build the final audio in memory with crossfades
    final_audio = np.array([], dtype=np.int16)
    first_params = None
    
    for i, input_file in enumerate(input_files):
        try:
            with wave.open(input_file, 'rb') as wav:
                if first_params is None:
                    first_params = wav.getparams()
                elif wav.getparams() != first_params:
                    print(f"Warning: WAV file {input_file} has different parameters")
                    
                frames = wav.readframes(wav.getnframes())
                audio = np.frombuffer(frames, dtype=np.int16)
                
                if i == 0:
                    # First segment - use as is
                    final_audio = audio
                else:
                    # Apply crossfade with previous segment
                    if len(final_audio) >= crossfade_samples and len(audio) >= crossfade_samples:
                        # Create crossfade weights
                        fade_out = np.linspace(1.0, 0.0, crossfade_samples)
                        fade_in = np.linspace(0.0, 1.0, crossfade_samples)
                        
                        # Apply crossfade
                        crossfade_region = (final_audio[-crossfade_samples:] * fade_out + 
                                           audio[:crossfade_samples] * fade_in).astype(np.int16)
                        
                        # Combine: original without last crossfade_samples + crossfade + new without first crossfade_samples
                        final_audio = np.concatenate([final_audio[:-crossfade_samples], 
                                                    crossfade_region, 
                                                    audio[crossfade_samples:]])
                    else:
                        # One segment too short for crossfade, just append
                        print(f"Segment {i} too short for crossfade, concatenating directly")
                        final_audio = np.concatenate([final_audio, audio])
        except Exception as e:
            print(f"Error processing file {input_file}: {e}")
            if i == 0:
                raise  # Critical failure if first file fails
    
    # Write the final audio data to the output file
    try:
        with wave.open(output_file, 'wb') as output_wav:
            if first_params is None:
                raise ValueError("No valid WAV files were processed")
                
            output_wav.setparams(first_params)
            output_wav.writeframes(final_audio.tobytes())
        
        print(f"Successfully stitched audio to {output_file} with crossfading")
    except Exception as e:
        print(f"Error writing output file {output_file}: {e}")
        raise

def list_available_voices():
    """List all available voices with the recommended one marked."""
    print("Available voices (in order of conversational realism):")
    for i, voice in enumerate(AVAILABLE_VOICES):
        marker = "★" if voice == DEFAULT_VOICE else " "
        print(f"{marker} {voice}")
    print(f"\nDefault voice: {DEFAULT_VOICE}")
    
    print("\nAvailable emotion tags:")
    print("<laugh>, <chuckle>, <sigh>, <cough>, <sniffle>, <groan>, <yawn>, <gasp>")

def main():
    # Parse command line arguments
    parser = argparse.ArgumentParser(description="Orpheus Text-to-Speech using Orpheus-FASTAPI")
    parser.add_argument("--text", type=str, help="Text to convert to speech")
    parser.add_argument("--voice", type=str, default=DEFAULT_VOICE, help=f"Voice to use (default: {DEFAULT_VOICE})")
    parser.add_argument("--output", type=str, help="Output WAV file path")
    parser.add_argument("--list-voices", action="store_true", help="List available voices")
    parser.add_argument("--temperature", type=float, default=TEMPERATURE, help="Temperature for generation")
    parser.add_argument("--top_p", type=float, default=TOP_P, help="Top-p sampling parameter")
    parser.add_argument("--repetition_penalty", type=float, default=REPETITION_PENALTY, 
                       help="Repetition penalty (fixed at 1.1 for stable generation - parameter kept for compatibility)")
    
    args = parser.parse_args()
    
    if args.list_voices:
        list_available_voices()
        return
    
    # Use text from command line or prompt user
    prompt = args.text
    if not prompt:
        if len(sys.argv) > 1 and sys.argv[1] not in ("--voice", "--output", "--temperature", "--top_p", "--repetition_penalty"):
            prompt = " ".join([arg for arg in sys.argv[1:] if not arg.startswith("--")])
        else:
            prompt = input("Enter text to synthesize: ")
            if not prompt:
                prompt = "Hello, I am Orpheus, an AI assistant with emotional speech capabilities."
    
    # Default output file if none provided
    output_file = args.output
    if not output_file:
        # Create outputs directory if it doesn't exist
        os.makedirs("outputs", exist_ok=True)
        # Generate a filename based on the voice and a timestamp
        timestamp = time.strftime("%Y%m%d_%H%M%S")
        output_file = f"outputs/{args.voice}_{timestamp}.wav"
        print(f"No output file specified. Saving to {output_file}")
    
    # Generate speech
    start_time = time.time()
    audio_segments = generate_speech_from_api(
        prompt=prompt,
        voice=args.voice,
        temperature=args.temperature,
        top_p=args.top_p,
        repetition_penalty=args.repetition_penalty,
        output_file=output_file
    )
    end_time = time.time()
    
    print(f"Speech generation completed in {end_time - start_time:.2f} seconds")
    print(f"Audio saved to {output_file}")

if __name__ == "__main__":
    main()