diff --git a/config.json b/config.json
new file mode 100644
index 0000000..1fb7482
--- /dev/null
+++ b/config.json
@@ -0,0 +1,12 @@
+{
+    "model": "mlx-community/whisper-base-mlx",
+    "device": null,
+    "lang": null,
+    "silence": 1000,
+    "max_buffer": 20,
+    "channels": 1,
+    "es": false,
+    "fr": false,
+    "ar": false,
+    "ingest": false
+}
diff --git a/engine_distribute.py b/engine_distribute.py
new file mode 100644
index 0000000..69f4a83
--- /dev/null
+++ b/engine_distribute.py
@@ -0,0 +1,49 @@
+import requests
+import time
+import argparse
+import multiprocessing
+
+INGEST_URL = "https://emiapi.reynafamily.com/live-captions/ingest"
+
+def run_distribution(in_queue, args):
+    print("[Distribute] Starting distribution service...")
+    
+    while True:
+        try:
+            payload = in_queue.get()
+            if payload is None: break
+            
+            if args.ingest:
+                delay = 1
+                max_delay = 15
+                success = False
+                
+                while not success:
+                    try:
+                        response = requests.post(INGEST_URL, json=payload, timeout=5)
+                        if response.status_code == 200:
+                            success = True
+                        else:
+                            print(f"[Distribute Error] {response.status_code}. Retrying in {delay}s...")
+                    except Exception as e:
+                        print(f"[Distribute Error] {e}. Retrying in {delay}s...")
+                    
+                    if not success:
+                        time.sleep(delay)
+                        delay = min(delay * 2, max_delay)
+                
+                print(f"[Distribute] Sent payload: {list(payload.keys())}")
+            else:
+                print(f"[Distribute] Local display (ingest disabled): {payload}")
+                
+        except Exception as e:
+            print(f"[Distribute] Error: {e}")
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("-i", "--ingest", action="store_true")
+    args = parser.parse_args()
+    
+    # Dummy queue for testing
+    in_q = multiprocessing.Queue()
+    run_distribution(in_q, args)
diff --git a/engine_transcribe.py b/engine_transcribe.py
new file mode 100644
index 0000000..6aa1172
--- /dev/null
+++ b/engine_transcribe.py
@@ -0,0 +1,143 @@
+import sys
+from unittest.mock import MagicMock
+
+# Comprehensive workaround for missing _lzma in some Python builds
+try:
+    import lzma
+except ImportError:
+    mock_lzma = MagicMock()
+    mock_lzma.FORMAT_XZ = 1
+    mock_lzma.FORMAT_ALONE = 2
+    mock_lzma.FORMAT_RAW = 3
+    mock_lzma.CHECK_NONE = 0
+    mock_lzma.CHECK_CRC32 = 1
+    mock_lzma.CHECK_CRC64 = 4
+    mock_lzma.CHECK_SHA256 = 10
+    sys.modules["_lzma"] = MagicMock()
+    sys.modules["lzma"] = mock_lzma
+
+import sys
+import time
+import numpy as np
+import sounddevice as sd
+import torch
+import mlx_whisper
+from silero_vad import load_silero_vad, get_speech_timestamps
+import queue
+import argparse
+
+# Audio Constants
+SAMPLERATE = 16000
+BLOCK_SIZE = 512
+VAD_THRESHOLD = 0.5
+
+def transcribe_with_controls(audio, transcribe_kwargs):
+    """Call mlx_whisper.transcribe with graceful fallback for unsupported args."""
+    try:
+        return mlx_whisper.transcribe(audio, **transcribe_kwargs)
+    except Exception as exc:
+        # Fallback logic for beam_size or other specific MLX implementation gaps
+        if "beam_size" in transcribe_kwargs:
+            transcribe_kwargs.pop("beam_size")
+            return mlx_whisper.transcribe(audio, **transcribe_kwargs)
+        raise exc
+
+def audio_callback(indata, frames, time, status, audio_queue):
+    if status:
+        print(status, file=sys.stderr)
+    audio_queue.put(indata.copy())
+
+def run_transcription(out_queue, args):
+    device = "mps" if torch.backends.mps.is_available() else "cpu"
+    print(f"[Transcribe] Loading Whisper model '{args.model}' on {device}...")
+    
+    vad_model = load_silero_vad()
+    audio_queue = queue.Queue()
+    
+    # Selection of device
+    device_index = args.device
+
+    def callback_wrapper(indata, frames, time, status):
+        audio_callback(indata, frames, time, status, audio_queue)
+
+    audio_buffer = []
+    speech_started = False
+    buffer_limit = SAMPLERATE * args.max_buffer
+    
+    print(f"[Transcribe] Starting audio stream on device {device_index}...")
+    
+    try:
+        with sd.InputStream(samplerate=SAMPLERATE, channels=args.channels, callback=callback_wrapper, blocksize=BLOCK_SIZE, device=device_index):
+            while True:
+                while not audio_queue.empty():
+                    data = audio_queue.get()
+                    if args.channels > 1:
+                        data = np.mean(data, axis=1) # Mix to mono
+                    audio_buffer.append(data.flatten())
+
+                if audio_buffer:
+                    current_audio = np.concatenate(audio_buffer)
+                    audio_tensor = torch.from_numpy(current_audio)
+                    buffer_duration = len(current_audio) / SAMPLERATE
+                    
+                    speech_timestamps = get_speech_timestamps(
+                        audio_tensor, 
+                        vad_model, 
+                        sampling_rate=SAMPLERATE,
+                        threshold=VAD_THRESHOLD,
+                        min_silence_duration_ms=args.silence
+                    )
+
+                    if speech_timestamps:
+                        speech_started = True
+                        last_end = speech_timestamps[-1]['end']
+                        buffer_len_samples = len(current_audio)
+                        
+                        if (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000) or buffer_len_samples > buffer_limit:
+                            # 1. Transcribe
+                            transcribe_kwargs = {
+                                "path_or_hf_repo": args.model,
+                                "temperature": (0.0, 0.2, 0.4, 0.6, 0.8, 1.0),
+                            }
+                            if args.lang:
+                                transcribe_kwargs["language"] = args.lang
+
+                            result = transcribe_with_controls(current_audio, transcribe_kwargs)
+                            text = result['text'].strip()
+                            detected_lang = result.get('language', 'en')
+
+                            if text:
+                                # Send to Translation Process
+                                out_queue.put({
+                                    "original": text,
+                                    "detected_lang": detected_lang,
+                                    "ts": time.time()
+                                })
+                                print(f"[Transcribe] {detected_lang.upper()}: {text}")
+
+                            audio_buffer = []
+                            speech_started = False
+                    
+                    elif not speech_started and len(current_audio) > SAMPLERATE * 2:
+                        audio_buffer = []
+
+    except KeyboardInterrupt:
+        print("[Transcribe] Stopped.")
+    except Exception as e:
+        print(f"[Transcribe] Error: {e}")
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--model", type=str, default="mlx-community/whisper-base-mlx")
+    parser.add_argument("--device", type=int, default=None)
+    parser.add_argument("--lang", type=str, default=None)
+    parser.add_argument("--silence", type=int, default=1000)
+    parser.add_argument("--max-buffer", type=int, default=20)
+    parser.add_argument("--channels", type=int, default=1)
+    args = parser.parse_args()
+    
+    # This part would normally be called by the main coordinator
+    # For testing, we can use a dummy queue
+    import multiprocessing
+    q = multiprocessing.Queue()
+    run_transcription(q, args)
diff --git a/engine_translate.py b/engine_translate.py
new file mode 100644
index 0000000..d24fc7e
--- /dev/null
+++ b/engine_translate.py
@@ -0,0 +1,130 @@
+import sys
+from unittest.mock import MagicMock
+
+# Comprehensive workaround for missing _lzma in some Python builds
+try:
+    import lzma
+except ImportError:
+    mock_lzma = MagicMock()
+    mock_lzma.FORMAT_XZ = 1
+    mock_lzma.FORMAT_ALONE = 2
+    mock_lzma.FORMAT_RAW = 3
+    mock_lzma.CHECK_NONE = 0
+    mock_lzma.CHECK_CRC32 = 1
+    mock_lzma.CHECK_CRC64 = 4
+    mock_lzma.CHECK_SHA256 = 10
+    sys.modules["_lzma"] = MagicMock()
+    sys.modules["lzma"] = mock_lzma
+
+import torch
+from transformers import MarianMTModel, MarianTokenizer
+import re
+import argparse
+import multiprocessing
+import time
+
+TARGET_LANGS = {
+    "es": "Helsinki-NLP/opus-mt-en-es",
+    "fr": "Helsinki-NLP/opus-mt-en-fr",
+    "ar": "Helsinki-NLP/opus-mt-en-ar"
+}
+
+def split_text_for_translation(text, max_chars=250):
+    normalized = " ".join(text.split()).strip()
+    if not normalized or len(normalized) <= max_chars:
+        return [normalized] if normalized else []
+
+    chunks = []
+    current = ""
+    sentences = [s for s in re.split(r"(?<=[.!?])\s+", normalized) if s]
+
+    for sentence in sentences:
+        if len(sentence) > max_chars:
+            words = sentence.split()
+            word_chunk = ""
+            for word in words:
+                candidate = f"{word_chunk} {word}".strip()
+                if len(candidate) <= max_chars:
+                    word_chunk = candidate
+                else:
+                    if word_chunk: chunks.append(word_chunk)
+                    word_chunk = word
+            if word_chunk: chunks.append(word_chunk)
+            continue
+
+        candidate = f"{current} {sentence}".strip()
+        if candidate and len(candidate) <= max_chars:
+            current = candidate
+        else:
+            if current: chunks.append(current)
+            current = sentence
+
+    if current: chunks.append(current)
+    return chunks
+
+def run_translation(in_queue, out_queue, args):
+    device = "mps" if torch.backends.mps.is_available() else "cpu"
+    print(f"[Translate] Loading translation models on {device}...")
+    
+    translation_engines = {}
+    for lang_key, model_id in TARGET_LANGS.items():
+        if getattr(args, lang_key, False):
+            print(f"[Translate] Loading {lang_key} model...")
+            tokenizer = MarianTokenizer.from_pretrained(model_id)
+            model = MarianMTModel.from_pretrained(model_id).to(device)
+            translation_engines[lang_key] = (model, tokenizer)
+
+    paragraph_buffer = []
+    
+    print("[Translate] Ready.")
+    
+    while True:
+        try:
+            item = in_queue.get()
+            if item is None: break
+            
+            text = item.get("original", "").strip()
+            detected_lang = item.get("detected_lang", "en")
+            
+            # Simple paragraph logic: 
+            # If the segment ends with sentence-terminal punctuation, flush the paragraph.
+            paragraph_buffer.append(text)
+            
+            if text.endswith((".", "?", "!")):
+                full_text = " ".join(paragraph_buffer)
+                payload = {"original": full_text, "en": full_text if detected_lang == "en" else None}
+                
+                # If detected language is not English, we'd normally bridge to English first.
+                # For now, let's assume direct translation for simplicity or bridge if needed.
+                # (Refining bridge logic can come later)
+                
+                for lang_key, (model, tokenizer) in translation_engines.items():
+                    chunks = split_text_for_translation(full_text)
+                    translated_parts = []
+                    for chunk in chunks:
+                        inputs = tokenizer(chunk, return_tensors="pt", padding=True).to(device)
+                        with torch.no_grad():
+                            translated_tokens = model.generate(**inputs, max_new_tokens=150)
+                        translated_parts.append(tokenizer.decode(translated_tokens[0], skip_special_tokens=True).strip())
+                    
+                    translated_text = " ".join(part for part in translated_parts if part).strip()
+                    payload[lang_key] = translated_text
+                    print(f"[Translate] {lang_key.upper()}: {translated_text}")
+
+                out_queue.put(payload)
+                paragraph_buffer = []
+                
+        except Exception as e:
+            print(f"[Translate] Error: {e}")
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("-es", action="store_true")
+    parser.add_argument("-fr", action="store_true")
+    parser.add_argument("-ar", action="store_true")
+    args = parser.parse_args()
+    
+    # Dummy queues for testing
+    in_q = multiprocessing.Queue()
+    out_q = multiprocessing.Queue()
+    run_translation(in_q, out_q, args)
diff --git a/main_v2.py b/main_v2.py
new file mode 100644
index 0000000..c14ac99
--- /dev/null
+++ b/main_v2.py
@@ -0,0 +1,84 @@
+import sys
+from unittest.mock import MagicMock
+
+# Comprehensive workaround for missing _lzma in some Python builds
+try:
+    import lzma
+except ImportError:
+    mock_lzma = MagicMock()
+    mock_lzma.FORMAT_XZ = 1
+    mock_lzma.FORMAT_ALONE = 2
+    mock_lzma.FORMAT_RAW = 3
+    mock_lzma.CHECK_NONE = 0
+    mock_lzma.CHECK_CRC32 = 1
+    mock_lzma.CHECK_CRC64 = 4
+    mock_lzma.CHECK_SHA256 = 10
+    sys.modules["_lzma"] = MagicMock()
+    sys.modules["lzma"] = mock_lzma
+
+import multiprocessing
+import argparse
+import sys
+import json
+import os
+from engine_transcribe import run_transcription
+from engine_translate import run_translation
+from engine_distribute import run_distribution
+
+def main():
+    config_path = "config.json"
+    defaults = {}
+    if os.path.exists(config_path):
+        with open(config_path, "r") as f:
+            defaults = json.load(f)
+
+    parser = argparse.ArgumentParser(description="Multi-process Transcription and Translation.")
+    
+    # Transcribe Args
+    parser.add_argument("--model", type=str, default=defaults.get("model", "mlx-community/whisper-base-mlx"))
+    parser.add_argument("--device", type=int, default=defaults.get("device"))
+    parser.add_argument("--lang", type=str, default=defaults.get("lang"))
+    parser.add_argument("--silence", type=int, default=defaults.get("silence", 1000))
+    parser.add_argument("--max-buffer", type=int, default=defaults.get("max_buffer", 20))
+    parser.add_argument("--channels", type=int, default=defaults.get("channels", 1))
+    
+    # Translate Args
+    parser.add_argument("-es", action="store_true", default=defaults.get("es", False), help="Enable Spanish translation")
+    parser.add_argument("-fr", action="store_true", default=defaults.get("fr", False), help="Enable French translation")
+    parser.add_argument("-ar", action="store_true", default=defaults.get("ar", False), help="Enable Arabic translation")
+    
+    # Distribute Args
+    parser.add_argument("-i", "--ingest", action="store_true", default=defaults.get("ingest", False), help="Enable data transmission to server")
+    
+    args = parser.parse_args()
+
+    # Queues for communication
+    # Transcribe -> Translate
+    q_trans_to_tl = multiprocessing.Queue()
+    # Translate -> Distribute
+    q_tl_to_dist = multiprocessing.Queue()
+
+    # Processes
+    p_transcribe = multiprocessing.Process(target=run_transcription, args=(q_trans_to_tl, args))
+    p_translate = multiprocessing.Process(target=run_translation, args=(q_trans_to_tl, q_tl_to_dist, args))
+    p_distribute = multiprocessing.Process(target=run_distribution, args=(q_tl_to_dist, args))
+
+    print("[Main] Starting processes...")
+    p_transcribe.start()
+    p_translate.start()
+    p_distribute.start()
+
+    try:
+        p_transcribe.join()
+        p_translate.join()
+        p_distribute.join()
+    except KeyboardInterrupt:
+        print("\n[Main] Stopping processes...")
+        p_transcribe.terminate()
+        p_translate.terminate()
+        p_distribute.terminate()
+        sys.exit(0)
+
+if __name__ == "__main__":
+    multiprocessing.freeze_support()
+    main()
diff --git a/requirements.txt b/requirements.txt
index fabf9c5..d94fe0a 100644
--- a/requirements.txt
+++ b/requirements.txt
@@ -9,3 +9,7 @@ pyinstaller
 sacremoses
 joblib
 sentencepiece
+pyannote.audio
+soundfile
+langdetect
+python-dotenv
diff --git a/transcribe.py b/transcribe.py
index d689cb0..5ef2fa1 100644
--- a/transcribe.py
+++ b/transcribe.py
@@ -33,6 +33,18 @@ import queue
 import torch
 from silero_vad import load_silero_vad, get_speech_timestamps
 from transformers import MarianMTModel, MarianTokenizer
+from pyannote.audio import Pipeline
+from dotenv import load_dotenv
+from langdetect import detect, DetectorFactory
+from langdetect.lang_detect_exception import LangDetectException
+
+# Enforce deterministic langdetect results
+DetectorFactory.seed = 0
+
+
+# Load environment variables from .env file
+load_dotenv()
+
 
 # Parameters
 WHISPER_MODEL = "mlx-community/whisper-base-mlx" 
@@ -168,51 +180,6 @@ def quality_score(mean_logprob, max_compression):
     compression_penalty = 0.25 * max(0.0, (max_compression or 1.5) - 1.5)
     return mean_logprob - compression_penalty
 
-def cosine_similarity(vec_a, vec_b):
-    if vec_a is None or vec_b is None:
-        return None
-    denom = (np.linalg.norm(vec_a) * np.linalg.norm(vec_b))
-    if denom == 0:
-        return None
-    return float(np.dot(vec_a, vec_b) / denom)
-
-def compute_speaker_signature(audio, samplerate):
-    """Compute a lightweight speaker signature from averaged log spectrum."""
-    if audio is None or len(audio) < int(0.5 * samplerate):
-        return None
-
-    data = np.asarray(audio, dtype=np.float32)
-    if data.ndim > 1:
-        data = np.mean(data, axis=1)
-
-    # Keep only voiced-ish samples to reduce silence/noise influence.
-    voiced = data[np.abs(data) > 0.01]
-    if len(voiced) < int(0.3 * samplerate):
-        voiced = data
-
-    frame_size = 1024
-    hop = 512
-    if len(voiced) < frame_size:
-        return None
-
-    window = np.hanning(frame_size).astype(np.float32)
-    spectra = []
-    for i in range(0, len(voiced) - frame_size + 1, hop):
-        frame = voiced[i:i + frame_size] * window
-        mag = np.abs(np.fft.rfft(frame))
-        if mag.size > 1:
-            mag = mag[1:257]  # Focus on lower-frequency envelope.
-            spectra.append(np.log1p(mag))
-
-    if not spectra:
-        return None
-
-    signature = np.mean(np.stack(spectra, axis=0), axis=0)
-    norm = np.linalg.norm(signature)
-    if norm == 0:
-        return None
-    return signature / norm
-
 def should_retry_transcription(text, mean_logprob, max_compression, args):
     if not text:
         return False
@@ -373,6 +340,34 @@ def post_correct_with_local_llm(text, args, llm_state, context_lines=None):
             print(f"[POST-LLM][ERROR]: {exc}", flush=True)
     return text
 
+def structure_paragraph_with_llm(text, args, llm_state):
+    """Use a local LLM to structure a series of captions into a paragraph."""
+    prompt = (
+        "You are a real-time English caption editor. Your task is to take a series of transcribed audio segments and format them into a single, coherent paragraph.\n"
+        "Rules:\n"
+        "1. Combine the segments into a flowing paragraph.\n"
+        "2. Correct grammar, punctuation, and casing.\n"
+        "3. Remove disfluencies and false starts (e.g., 'um', 'uh').\n"
+        "4. Do not change the meaning of the text.\n"
+        "5. If there are speaker labels (e.g., [SPEAKER_01]:), preserve them.\n"
+        "6. Output ONLY the formatted paragraph.\n\n"
+        "Here are the segments:\n"
+        f"{text}\n\n"
+        "Formatted paragraph:"
+    )
+
+    try:
+        response = call_ollama_generate(args, prompt, args.post_correct_llm_timeout, temperature=0.2)
+        response.raise_for_status()
+        raw = response.json().get("response", "").strip()
+        return raw
+    except Exception as exc:
+        if not llm_state.get("paragraph_warned", False):
+            print(f"\n[SYSTEM]: Local paragraph structuring LLM unavailable ({exc}).")
+            llm_state["paragraph_warned"] = True
+        return text # Fallback to the original text
+
+
 def maybe_merge_recent_caption(recent_en_lines, current_text, now_ts, window_sec):
     if not recent_en_lines:
         return None
@@ -507,6 +502,38 @@ def is_hallucination(text):
             
     return False
 
+def assign_speakers_to_segments(segments, diarization):
+    """Assign speaker labels to transcription segments based on diarization."""
+    if not diarization or not segments:
+        return segments
+    
+    for segment in segments:
+        segment_start = segment['start']
+        segment_end = segment['end']
+        
+        speaker_intersections = {}
+        for turn, _, speaker in diarization.itertracks(yield_label=True):
+            intersection_start = max(segment_start, turn.start)
+            intersection_end = min(segment_end, turn.end)
+            
+            if intersection_end > intersection_start:
+                intersection_duration = intersection_end - intersection_start
+                if speaker not in speaker_intersections:
+                    speaker_intersections[speaker] = 0
+                speaker_intersections[speaker] += intersection_duration
+        
+        if speaker_intersections:
+            dominant_speaker = max(speaker_intersections, key=speaker_intersections.get)
+            segment['speaker'] = dominant_speaker
+        else:
+            # Keep speaker from previous segment if segment is short and follows closely
+            # This can help with short utterances that are missed by the diarizer
+            # For now, we'll just label as unknown
+            segment['speaker'] = 'UNKNOWN'
+            
+    return segments
+
+
 def callback(indata, frames, time, status):
     if status:
         print(status, file=sys.stderr)
@@ -569,6 +596,7 @@ def main():
     parser.add_argument("-s", "--stream", action="store_true", help="Enable real-time streaming transcription (Draft mode)")
     parser.add_argument("-c", "--context", action="store_true", help="Enable prompt caching/rolling context for better continuity")
     parser.add_argument("--lang", type=str, help="Hardcode source language (e.g. 'en', 'es') to bypass detection")
+    parser.add_argument("--detect-auto-translation", action="store_true", help="Enable heuristic to detect and leverage auto-translated English segments.")
     parser.add_argument("--silence", type=int, default=1000, help="Minimum silence duration in ms to end a chunk (default: 1000)")
     parser.add_argument("--max-buffer", type=int, default=20, help="Maximum buffer duration in seconds before forcing a flush (default: 20)")
     parser.add_argument("--channels", type=int, default=1, help="Number of input channels (default: 1)")
@@ -594,7 +622,7 @@ def main():
     parser.add_argument("--glossary-pair", action="append", type=parse_glossary_pair, default=[], help="Term replacement pair SOURCE=TARGET (repeatable)")
     parser.add_argument("--post-correct", action="store_true", help="Enable v2 finalized English caption post-correction")
     parser.add_argument("--post-correct-llm", action="store_true", help="Use local LLM for post-correction (fallback to rules on failure)")
-    parser.add_argument("--post-correct-model", type=str, default="llama3.1:8b-instruct", help="Local Ollama model for post-correction")
+    parser.add_argument("--post-correct-model", type=str, default="qwen:2b", help="Local Ollama model for post-correction")
     parser.add_argument("--post-correct-ollama-url", type=str, default="http://127.0.0.1:11434/api/generate", help="Ollama generate endpoint for local post-correction")
     parser.add_argument("--post-correct-llm-timeout", type=float, default=8.0, help="Timeout seconds for local LLM post-correction (default: 8.0)")
     parser.add_argument("--post-correct-keep-alive", type=str, default="30m", help="Ollama keep_alive duration to keep the model loaded (default: 30m)")
@@ -602,12 +630,10 @@ def main():
     parser.add_argument("--skip-post-correct-warmup", action="store_true", help="Skip startup warmup call for the post-correct LLM")
     parser.add_argument("--post-correct-min-overlap", type=float, default=0.45, help="Minimum token overlap ratio required to accept LLM rewrite (default: 0.45)")
     parser.add_argument("--post-correct-debug", action="store_true", help="Print pre/post LLM correction text for debugging")
+    parser.add_argument("--llm-paragraph", action="store_true", help="Enable LLM-based paragraph structuring.")
     parser.add_argument("--llm-merge-decider", action="store_true", help="Use local LLM to validate/refine smart-correct line merges")
     parser.add_argument("--llm-merge-timeout", type=float, default=1.5, help="Timeout seconds for LLM merge decision (default: 1.5)")
-    parser.add_argument("--speaker-change-detect", action="store_true", help="Enable lightweight speaker-change detection to force caption line cuts")
-    parser.add_argument("--speaker-sim-threshold", type=float, default=0.72, help="Cosine similarity threshold below which a speaker change is assumed (default: 0.72)")
-    parser.add_argument("--speaker-min-buffer", type=float, default=1.6, help="Minimum buffered speech seconds before speaker-change checks (default: 1.6)")
-    parser.add_argument("--speaker-check-interval", type=float, default=0.6, help="Seconds between speaker-change checks while buffering (default: 0.6)")
+    parser.add_argument("--speaker-diarization", action="store_true", help="Enable speaker diarization using pyannote.audio.")
     parser.add_argument("--session-log-file", type=str, default="transcribe_session.log", help="Path to session debug log file (default: transcribe_session.log)")
     parser.add_argument("--disable-session-log", action="store_true", help="Disable writing session debug logs to file")
     
@@ -646,15 +672,6 @@ def main():
         print("[SYSTEM]: --llm-merge-decider requires --post-correct-llm. Falling back to heuristic merge.")
         args.llm_merge_decider = False
         log_session("[SYSTEM]: --llm-merge-decider requires --post-correct-llm. Falling back to heuristic merge.")
-    if args.speaker_change_detect:
-        print(
-            "Speaker-change detection enabled "
-            f"(threshold={args.speaker_sim_threshold}, min_buffer={args.speaker_min_buffer}s)."
-        )
-        log_session(
-            "Speaker-change detection enabled "
-            f"(threshold={args.speaker_sim_threshold}, min_buffer={args.speaker_min_buffer}s)."
-        )
 
     if args.quantize:
         WHISPER_MODEL = "mlx-community/whisper-small-mlx-4bit"
@@ -670,6 +687,26 @@ def main():
     print(f"Using device: {device}")
     log_session(f"Using device: {device}")
 
+    diarization_pipeline = None
+    if args.speaker_diarization:
+        print("Loading speaker diarization pipeline...")
+        hf_token = os.environ.get("HF_TOKEN")
+        if not hf_token:
+            print("[WARN] Hugging Face token not found (HF_TOKEN env variable). Diarization may fail if model is private.")
+            log_session("[WARN] Hugging Face token not found (HF_TOKEN env variable).")
+        
+        try:
+            diarization_pipeline = Pipeline.from_pretrained(
+                "pyannote/speaker-diarization-3.1"
+            )
+            diarization_pipeline.to(torch.device(device))
+            print("Speaker diarization pipeline loaded.")
+            log_session("Speaker diarization pipeline loaded.")
+        except Exception as e:
+            print(f"[ERROR] Could not load speaker diarization pipeline: {e}")
+            log_session(f"[ERROR] Could not load speaker diarization pipeline: {e}")
+            diarization_pipeline = None
+
     # Only start ingest thread if enabled
     if args.ingest:
         threading.Thread(target=ingest_worker, daemon=True).start()
@@ -743,9 +780,8 @@ def main():
     last_draft_text = ""
     draft_line_active = False
     recent_en_lines = deque(maxlen=2)
-    llm_post_correct_state = {"available": True, "warned": False, "merge_warned": False}
-    speaker_reference_signature = None
-    last_speaker_check_time = 0.0
+    llm_post_correct_state = {"available": True, "warned": False, "merge_warned": False, "paragraph_warned": False}
+    english_paragraph_buffer = []
 
     try:
         with sd.InputStream(samplerate=SAMPLERATE, channels=args.channels, callback=callback, blocksize=BLOCK_SIZE, device=device_index):
@@ -785,7 +821,6 @@ def main():
                         rolling_context = ""
                         rolling_context_en = ""
                         recent_en_lines.clear()
-                        speaker_reference_signature = None
                         last_draft_text = ""
                         last_change_time = time.time()
                         if args.stream:
@@ -797,30 +832,10 @@ def main():
                         speech_started = True
                         last_end = speech_timestamps[-1]['end']
                         buffer_len_samples = len(current_audio)
-                        force_flush_for_speaker_change = False
-
-                        if args.speaker_change_detect and buffer_duration >= args.speaker_min_buffer:
-                            now = time.time()
-                            if (now - last_speaker_check_time) >= args.speaker_check_interval:
-                                last_speaker_check_time = now
-                                tail_secs = min(2.0, buffer_duration)
-                                tail_samples = int(tail_secs * SAMPLERATE)
-                                recent_signature = compute_speaker_signature(current_audio[-tail_samples:], SAMPLERATE)
-                                if speaker_reference_signature is None and recent_signature is not None:
-                                    speaker_reference_signature = recent_signature
-                                elif recent_signature is not None:
-                                    similarity = cosine_similarity(speaker_reference_signature, recent_signature)
-                                    if similarity is not None and similarity < args.speaker_sim_threshold:
-                                        force_flush_for_speaker_change = True
-                                        print(
-                                            f"\n[SYSTEM]: Speaker change detected (similarity={similarity:.2f}). "
-                                            "Cutting current line."
-                                        )
                         
                         if (
                             (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000)
                             or buffer_len_samples > buffer_limit
-                            or force_flush_for_speaker_change
                         ):
                             
                             # Clear draft line if it was used
@@ -830,7 +845,7 @@ def main():
                                 draft_line_active = False
 
                             # Prepare transcription kwargs
-                            transcribe_kwargs = {"path_or_hf_repo": WHISPER_MODEL}
+                            transcribe_kwargs = {"path_or_hf_repo": WHISPER_MODEL, "word_timestamps": args.speaker_diarization}
                             if args.lang:
                                 transcribe_kwargs["language"] = args.lang
                             if args.context and rolling_context:
@@ -841,9 +856,46 @@ def main():
                             transcribe_kwargs["logprob_threshold"] = args.logprob_threshold
                             transcribe_kwargs["compression_ratio_threshold"] = args.compression_threshold
 
+                            # Speaker diarization
+                            diarization_result = None
+                            if diarization_pipeline:
+                                log_session("Performing speaker diarization...")
+                                audio_for_diarization = torch.from_numpy(current_audio).float().unsqueeze(0)
+                                try:
+                                    diarization_result = diarization_pipeline({"waveform": audio_for_diarization, "sample_rate": SAMPLERATE})
+                                except Exception as e:
+                                    print(f"\n[ERROR] Diarization failed: {e}")
+                                    log_session(f"Diarization failed: {e}")
+
                             # 1. Transcribe & Detect Language
                             transcription_result = transcribe_with_controls(current_audio, transcribe_kwargs)
-                            original_text = transcription_result['text'].strip()
+                            
+                            original_text_for_print = ""
+                            if diarization_result:
+                                segments_with_speakers = assign_speakers_to_segments(
+                                    transcription_result.get('segments', []),
+                                    diarization_result
+                                )
+                                transcription_result['segments'] = segments_with_speakers
+                                
+                                # Reconstruct text with speaker labels for printing
+                                text_parts_for_print = []
+                                for i, segment in enumerate(segments_with_speakers):
+                                    speaker = segment.get('speaker', 'UNKNOWN')
+                                    text = segment['text'].strip()
+                                    
+                                    if i > 0 and speaker == segments_with_speakers[i-1].get('speaker', 'UNKNOWN'):
+                                        text_parts_for_print.append(f" {text}")
+                                    else:
+                                        if i > 0: text_parts_for_print.append("\n")
+                                        text_parts_for_print.append(f"[{speaker}]: {text}")
+                                original_text_for_print = "".join(text_parts_for_print)
+
+                                # Reconstruct text without speaker labels for processing
+                                original_text = " ".join([seg['text'].strip() for seg in transcription_result['segments']])
+                            else:
+                                original_text = transcription_result['text'].strip()
+                                original_text_for_print = original_text
                             detected_lang = transcription_result.get('language', args.lang if args.lang else 'en')
 
                             avg_logprob, max_compression = extract_whisper_quality(transcription_result)
@@ -873,10 +925,20 @@ def main():
                                 rolling_context = ""
                                 rolling_context_en = ""
                                 recent_en_lines.clear()
-                                speaker_reference_signature = None
 
                             if original_text:
-                                print_caption(detected_lang, original_text, leading_newline=True)
+                                if args.detect_auto_translation and args.lang and args.lang != 'en' and len(original_text.split()) > 3:
+                                    try:
+                                        detected_lang_for_segment = detect(original_text)
+                                        if detected_lang_for_segment == 'en':
+                                            print_caption("AUTO-EN", original_text, leading_newline=True)
+                                            if args.context:
+                                                rolling_context_en = (rolling_context_en + " " + original_text)[-200:].strip()
+                                    except LangDetectException:
+                                        # Could not detect language, probably too short
+                                        pass
+                                
+                                print_caption(detected_lang, original_text_for_print, leading_newline=True)
                                 last_change_time = time.time() # Successfully transcribed full segment
 
                                 # Prepare payload
@@ -957,10 +1019,44 @@ def main():
                                     else:
                                         recent_en_lines.append({"text": english_for_caption, "ts": now_ts})
 
+                                if args.llm_paragraph:
+                                    english_paragraph_buffer.append(english_for_caption)
+                                    
+                                    if english_for_caption.strip().endswith((".", "?", "!")):
+                                        full_text = "\n".join(english_paragraph_buffer)
+                                        structured_paragraph = structure_paragraph_with_llm(full_text, args, llm_post_correct_state)
+                                        print(f"\n[LLM Paragraph]:\n{structured_paragraph}")
+                                        log_session(f"[LLM-P]: {structured_paragraph}")
+                                        
+                                        # Translate the paragraph here
+                                        if structured_paragraph and translation_engines:
+                                            english_chunks = split_text_for_translation(structured_paragraph, max_chars=args.mt_max_chars)
+                                            for lang_key, (model, tokenizer) in translation_engines.items():
+                                                translated_parts = []
+                                                for chunk in english_chunks:
+                                                    inputs = tokenizer(chunk, return_tensors="pt", padding=True).to(device)
+                                                    with torch.no_grad():
+                                                        translated_tokens = model.generate(
+                                                            **inputs,
+                                                            max_new_tokens=args.mt_max_new_tokens,
+                                                            num_beams=args.mt_num_beams,
+                                                            no_repeat_ngram_size=args.mt_no_repeat_ngram_size,
+                                                            length_penalty=args.mt_length_penalty,
+                                                            repetition_penalty=args.mt_repetition_penalty,
+                                                            early_stopping=not args.mt_no_early_stopping,
+                                                        )
+                                                    translated_parts.append(tokenizer.decode(translated_tokens[0], skip_special_tokens=True).strip())
+                                                translated_text = " ".join(part for part in translated_parts if part).strip()
+                                                print_caption(lang_key, translated_text)
+
+                                        english_paragraph_buffer = [] # Reset buffer
+                                
                                 if args.en and english_for_caption:
                                     payload["en"] = english_for_caption
-                                    print(f"[EN]: {english_for_caption}")
-                                    log_session(f"[EN]: {english_for_caption}")
+                                    # When llm_paragraph is enabled, we don't print the line-by-line English captions.
+                                    if not args.llm_paragraph:
+                                        print(f"[EN]: {english_for_caption}")
+                                        log_session(f"[EN]: {english_for_caption}")
 
                                 # Update rolling context for next segment
                                 if args.context:
@@ -968,7 +1064,7 @@ def main():
                                     rolling_context = (rolling_context + " " + original_text)[-200:].strip()
 
                                 # 3. Translate from English to other languages
-                                if english_for_translation and translation_engines:
+                                if not args.llm_paragraph and english_for_translation and translation_engines:
                                     english_chunks = split_text_for_translation(english_for_translation, max_chars=args.mt_max_chars)
                                     
                                     for lang_key, (model, tokenizer) in translation_engines.items():
@@ -1007,7 +1103,6 @@ def main():
                             last_stream_time = time.time()
                             stuck_draft_count = 0
                             last_draft_text = ""
-                            speaker_reference_signature = None
                             draft_line_active = False
 
                         elif args.stream and (time.time() - last_stream_time) > 1.0:
@@ -1064,7 +1159,6 @@ def main():
                     
                     elif not speech_started and len(current_audio) > SAMPLERATE * 2:
                         audio_buffer = []
-                        speaker_reference_signature = None
 
     except KeyboardInterrupt:
         print("\nStopped by user.")