diff --git a/requirements.txt b/requirements.txt
new file mode 100644
index 0000000..ef65724
--- /dev/null
+++ b/requirements.txt
@@ -0,0 +1,5 @@
+faster-whisper
+pyannote.audio
+python-dotenv
+pydub
+tqdm
diff --git a/transcriber.py b/transcriber.py
new file mode 100644
index 0000000..9ede4a0
--- /dev/null
+++ b/transcriber.py
@@ -0,0 +1,4 @@
+from transcriber.main import main
+
+if __name__ == "__main__":
+    main()
diff --git a/transcriber/__init__.py b/transcriber/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/asr/__init__.py b/transcriber/asr/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/asr/whisper_engine.py b/transcriber/asr/whisper_engine.py
new file mode 100644
index 0000000..aa4e56c
--- /dev/null
+++ b/transcriber/asr/whisper_engine.py
@@ -0,0 +1,93 @@
+import logging
+from dataclasses import dataclass, field
+
+from faster_whisper import WhisperModel
+
+logger = logging.getLogger(__name__)
+
+
+@dataclass
+class WordInfo:
+    """Single word with timing and confidence."""
+
+    word: str
+    start: float
+    end: float
+    probability: float
+
+
+@dataclass
+class Segment:
+    """ASR segment with text, timing, and optional word-level detail."""
+
+    start: float
+    end: float
+    text: str
+    words: list[WordInfo] = field(default_factory=list)
+
+
+class WhisperEngine:
+    """Speech recognition engine based on faster-whisper."""
+
+    def __init__(self, model_name: str, device: str, compute_type: str):
+        logger.info("Loading Whisper model: %s on %s (%s)", model_name, device, compute_type)
+        self._model = WhisperModel(model_name, device=device, compute_type=compute_type)
+
+    def transcribe(
+        self,
+        audio_path: str,
+        language: str | None = None,
+        beam_size: int = 5,
+        vad_filter: bool = True,
+    ) -> list[Segment]:
+        """Transcribe audio file and return list of segments.
+
+        Args:
+            audio_path: Path to WAV file.
+            language: Language code or None for auto-detection.
+            beam_size: Beam search width.
+            vad_filter: Whether to enable VAD filtering.
+
+        Returns:
+            List of transcription segments with word-level timestamps.
+        """
+        logger.info("Transcribing: %s", audio_path)
+
+        segments_gen, info = self._model.transcribe(
+            audio_path,
+            language=language,
+            beam_size=beam_size,
+            word_timestamps=True,
+            vad_filter=vad_filter,
+            vad_parameters={"min_silence_duration_ms": 500},
+            temperature=0.0,
+            condition_on_previous_text=False,
+            no_speech_threshold=0.6,
+            log_prob_threshold=-1.0,
+        )
+
+        logger.info(
+            "Detected language: %s (%.2f), duration: %.1fs",
+            info.language, info.language_probability, info.duration,
+        )
+
+        results = []
+        for seg in segments_gen:
+            words = [
+                WordInfo(
+                    word=w.word.strip(),
+                    start=w.start,
+                    end=w.end,
+                    probability=w.probability,
+                )
+                for w in (seg.words or [])
+            ]
+            results.append(Segment(
+                start=seg.start,
+                end=seg.end,
+                text=seg.text.strip(),
+                words=words,
+            ))
+
+        logger.info("Transcription complete: %d segments", len(results))
+        return results
diff --git a/transcriber/audio/__init__.py b/transcriber/audio/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/audio/chunking.py b/transcriber/audio/chunking.py
new file mode 100644
index 0000000..22f4a23
--- /dev/null
+++ b/transcriber/audio/chunking.py
@@ -0,0 +1,82 @@
+import logging
+import subprocess
+from dataclasses import dataclass
+from pathlib import Path
+
+logger = logging.getLogger(__name__)
+
+
+@dataclass
+class ChunkInfo:
+    """Metadata for a single audio chunk."""
+
+    path: str
+    start_offset: float
+    duration: float
+
+
+def get_audio_duration(wav_path: str) -> float:
+    """Get duration of audio file in seconds using ffprobe."""
+    cmd = [
+        "ffprobe", "-v", "quiet",
+        "-show_entries", "format=duration",
+        "-of", "csv=p=0",
+        wav_path,
+    ]
+    result = subprocess.run(cmd, capture_output=True, text=True, timeout=30)
+    if result.returncode != 0:
+        raise RuntimeError(f"ffprobe failed: {result.stderr[:300]}")
+    return float(result.stdout.strip())
+
+
+def chunk_audio(wav_path: str, max_duration_sec: int = 1800) -> list[ChunkInfo]:
+    """Split audio into chunks if longer than max_duration_sec.
+
+    Args:
+        wav_path: Path to the preprocessed WAV file.
+        max_duration_sec: Maximum chunk duration in seconds (default 30 min).
+
+    Returns:
+        List of ChunkInfo with paths and timing metadata.
+    """
+    total_duration = get_audio_duration(wav_path)
+    logger.info("Audio duration: %.1f sec", total_duration)
+
+    if total_duration <= max_duration_sec:
+        return [ChunkInfo(path=wav_path, start_offset=0.0, duration=total_duration)]
+
+    chunks = []
+    src = Path(wav_path)
+    chunk_dir = src.parent / "chunks"
+    chunk_dir.mkdir(exist_ok=True)
+
+    offset = 0.0
+    idx = 0
+    while offset < total_duration:
+        chunk_path = str(chunk_dir / f"{src.stem}_chunk{idx:03d}.wav")
+        remaining = total_duration - offset
+        duration = min(max_duration_sec, remaining)
+
+        cmd = [
+            "ffmpeg", "-y",
+            "-ss", str(offset),
+            "-i", wav_path,
+            "-t", str(duration),
+            "-c", "copy",
+            chunk_path,
+        ]
+        result = subprocess.run(cmd, capture_output=True, text=True, timeout=300)
+        if result.returncode != 0:
+            raise RuntimeError(f"Chunk {idx} failed: {result.stderr[:300]}")
+
+        chunks.append(ChunkInfo(
+            path=chunk_path,
+            start_offset=offset,
+            duration=duration,
+        ))
+        logger.info("Chunk %d: %.1fs - %.1fs", idx, offset, offset + duration)
+
+        offset += duration
+        idx += 1
+
+    return chunks
diff --git a/transcriber/audio/preprocess.py b/transcriber/audio/preprocess.py
new file mode 100644
index 0000000..4a5c48e
--- /dev/null
+++ b/transcriber/audio/preprocess.py
@@ -0,0 +1,54 @@
+import logging
+import subprocess
+from pathlib import Path
+
+logger = logging.getLogger(__name__)
+
+SUPPORTED_FORMATS = {".m4a", ".mp3", ".wav", ".aac"}
+
+
+def preprocess_audio(input_path: str, output_dir: str) -> str:
+    """Convert audio to mono 16kHz PCM WAV with normalization and DC offset removal.
+
+    Args:
+        input_path: Path to the source audio file.
+        output_dir: Directory for the processed file.
+
+    Returns:
+        Path to the processed WAV file.
+
+    Raises:
+        FileNotFoundError: If input file does not exist.
+        ValueError: If file format is not supported.
+        RuntimeError: If ffmpeg processing fails.
+    """
+    src = Path(input_path)
+    if not src.exists():
+        raise FileNotFoundError(f"Audio file not found: {input_path}")
+    if src.suffix.lower() not in SUPPORTED_FORMATS:
+        raise ValueError(
+            f"Unsupported format: {src.suffix}. Supported: {SUPPORTED_FORMATS}"
+        )
+
+    out = Path(output_dir) / f"{src.stem}_processed.wav"
+    out.parent.mkdir(parents=True, exist_ok=True)
+
+    cmd = [
+        "ffmpeg", "-y", "-i", str(src),
+        "-ac", "1",
+        "-ar", "16000",
+        "-sample_fmt", "s16",
+        "-af", "highpass=f=10,loudnorm=I=-16:TP=-1.5:LRA=11",
+        str(out),
+    ]
+
+    logger.info("Preprocessing: %s -> %s", src.name, out.name)
+
+    result = subprocess.run(
+        cmd, capture_output=True, text=True, timeout=600
+    )
+    if result.returncode != 0:
+        raise RuntimeError(f"ffmpeg failed: {result.stderr[:500]}")
+
+    logger.info("Preprocessing complete: %s", out.name)
+    return str(out)
diff --git a/transcriber/config.py b/transcriber/config.py
new file mode 100644
index 0000000..d3ea825
--- /dev/null
+++ b/transcriber/config.py
@@ -0,0 +1,38 @@
+import os
+from dataclasses import dataclass, field
+from pathlib import Path
+
+from dotenv import load_dotenv
+
+
+@dataclass
+class TranscriberConfig:
+    """Configuration for the transcription pipeline."""
+
+    input_path: str = ""
+    output_dir: str = "./output"
+    model: str = "large-v3"
+    device: str = "cuda"
+    compute_type: str = "float16"
+    language: str = "ru"
+    beam_size: int = 5
+    vad: bool = True
+    max_speakers: int | None = None
+    min_speakers: int | None = None
+    formats: list[str] = field(default_factory=lambda: ["txt", "json"])
+    pause_threshold: float = 1.5
+    chunk_duration: int = 1800
+    hf_token: str = ""
+
+    def __post_init__(self):
+        load_dotenv()
+        if not self.hf_token:
+            self.hf_token = os.getenv("HF_TOKEN", "")
+        if not self.hf_token:
+            raise ValueError(
+                "HF_TOKEN is required for pyannote diarization. "
+                "Set it in .env or pass via --hf-token"
+            )
+        if self.device == "cpu":
+            self.compute_type = "int8"
+        Path(self.output_dir).mkdir(parents=True, exist_ok=True)
diff --git a/transcriber/diarization/__init__.py b/transcriber/diarization/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/diarization/pyannote_engine.py b/transcriber/diarization/pyannote_engine.py
new file mode 100644
index 0000000..7ba07c1
--- /dev/null
+++ b/transcriber/diarization/pyannote_engine.py
@@ -0,0 +1,69 @@
+import logging
+from dataclasses import dataclass
+
+import torch
+from pyannote.audio import Pipeline
+from pyannote.audio.pipelines.utils.hook import ProgressHook
+
+logger = logging.getLogger(__name__)
+
+
+@dataclass
+class SpeakerTurn:
+    """A single speaker turn with timing."""
+
+    start: float
+    end: float
+    speaker: str
+
+
+class DiarizationEngine:
+    """Speaker diarization engine based on pyannote.audio."""
+
+    def __init__(self, hf_token: str, device: str):
+        logger.info("Loading diarization pipeline on %s", device)
+        self._pipeline = Pipeline.from_pretrained(
+            "pyannote/speaker-diarization-3.1",
+            token=hf_token,
+        )
+        self._device = torch.device(device)
+        self._pipeline.to(self._device)
+
+    def diarize(
+        self,
+        audio_path: str,
+        min_speakers: int | None = None,
+        max_speakers: int | None = None,
+    ) -> list[SpeakerTurn]:
+        """Run speaker diarization on audio file.
+
+        Args:
+            audio_path: Path to WAV file.
+            min_speakers: Minimum expected number of speakers.
+            max_speakers: Maximum expected number of speakers.
+
+        Returns:
+            List of speaker turns sorted by start time.
+        """
+        logger.info("Diarizing: %s", audio_path)
+
+        kwargs = {}
+        if min_speakers is not None:
+            kwargs["min_speakers"] = min_speakers
+        if max_speakers is not None:
+            kwargs["max_speakers"] = max_speakers
+
+        with ProgressHook() as hook:
+            diarization = self._pipeline(audio_path, hook=hook, **kwargs)
+
+        turns = []
+        for turn, _, speaker in diarization.itertracks(yield_label=True):
+            turns.append(SpeakerTurn(
+                start=turn.start,
+                end=turn.end,
+                speaker=speaker,
+            ))
+
+        speaker_set = {t.speaker for t in turns}
+        logger.info("Diarization complete: %d turns, %d speakers", len(turns), len(speaker_set))
+        return turns
diff --git a/transcriber/export/__init__.py b/transcriber/export/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/export/json_writer.py b/transcriber/export/json_writer.py
new file mode 100644
index 0000000..ff8f05c
--- /dev/null
+++ b/transcriber/export/json_writer.py
@@ -0,0 +1,34 @@
+import json
+from pathlib import Path
+
+from transcriber.merge.aligner import MergedSegment
+
+
+def write_json(segments: list[MergedSegment], output_path: str) -> str:
+    """Export merged segments as a structured JSON file.
+
+    Args:
+        segments: List of merged speaker segments.
+        output_path: Path to the output .json file.
+
+    Returns:
+        Path to the written file.
+    """
+    path = Path(output_path)
+    path.parent.mkdir(parents=True, exist_ok=True)
+
+    data = [
+        {
+            "speaker": seg.speaker,
+            "start": round(seg.start, 2),
+            "end": round(seg.end, 2),
+            "text": seg.text,
+        }
+        for seg in segments
+    ]
+
+    path.write_text(
+        json.dumps(data, ensure_ascii=False, indent=2),
+        encoding="utf-8",
+    )
+    return str(path)
diff --git a/transcriber/export/txt_writer.py b/transcriber/export/txt_writer.py
new file mode 100644
index 0000000..b97d24f
--- /dev/null
+++ b/transcriber/export/txt_writer.py
@@ -0,0 +1,26 @@
+from pathlib import Path
+
+from transcriber.merge.aligner import MergedSegment
+
+
+def write_txt(segments: list[MergedSegment], output_path: str) -> str:
+    """Export merged segments as a readable dialogue text file.
+
+    Args:
+        segments: List of merged speaker segments.
+        output_path: Path to the output .txt file.
+
+    Returns:
+        Path to the written file.
+    """
+    path = Path(output_path)
+    path.parent.mkdir(parents=True, exist_ok=True)
+
+    lines = []
+    for seg in segments:
+        lines.append(f"[{seg.speaker}]")
+        lines.append(seg.text)
+        lines.append("")
+
+    path.write_text("\n".join(lines), encoding="utf-8")
+    return str(path)
diff --git a/transcriber/main.py b/transcriber/main.py
new file mode 100644
index 0000000..7b31054
--- /dev/null
+++ b/transcriber/main.py
@@ -0,0 +1,70 @@
+import argparse
+import logging
+import sys
+
+from transcriber.config import TranscriberConfig
+from transcriber.pipeline import TranscriptionPipeline
+
+
+def parse_args() -> argparse.Namespace:
+    """Parse command-line arguments."""
+    parser = argparse.ArgumentParser(
+        description="Transcribe audio with speaker diarization",
+    )
+    parser.add_argument("input", help="Path to audio file (.m4a, .mp3, .wav, .aac)")
+    parser.add_argument("--output", default="./output", help="Output directory (default: ./output)")
+    parser.add_argument("--model", default="large-v3", help="Whisper model name (default: large-v3)")
+    parser.add_argument("--device", default="cuda", choices=["cuda", "cpu"], help="Device (default: cuda)")
+    parser.add_argument("--compute-type", default="float16", help="Compute type (default: float16)")
+    parser.add_argument("--language", default="ru", help="Language code (default: ru)")
+    parser.add_argument("--beam-size", type=int, default=5, help="Beam search size (default: 5)")
+    parser.add_argument("--vad", default="on", choices=["on", "off"], help="VAD filter (default: on)")
+    parser.add_argument("--max-speakers", type=int, default=None, help="Maximum number of speakers")
+    parser.add_argument("--min-speakers", type=int, default=None, help="Minimum number of speakers")
+    parser.add_argument("--format", nargs="+", default=["txt", "json"], help="Output formats (default: txt json)")
+    parser.add_argument("--pause-threshold", type=float, default=1.5, help="Max pause for merging (default: 1.5s)")
+    parser.add_argument("--chunk-duration", type=int, default=1800, help="Max chunk duration in sec (default: 1800)")
+    parser.add_argument("--hf-token", default="", help="HuggingFace token (default: from .env)")
+    parser.add_argument("--verbose", "-v", action="store_true", help="Enable debug logging")
+    return parser.parse_args()
+
+
+def main() -> None:
+    """Entry point for the transcription CLI."""
+    args = parse_args()
+
+    logging.basicConfig(
+        level=logging.DEBUG if args.verbose else logging.INFO,
+        format="%(asctime)s [%(levelname)s] %(name)s: %(message)s",
+        datefmt="%H:%M:%S",
+    )
+
+    config = TranscriberConfig(
+        input_path=args.input,
+        output_dir=args.output,
+        model=args.model,
+        device=args.device,
+        compute_type=args.compute_type,
+        language=args.language,
+        beam_size=args.beam_size,
+        vad=args.vad == "on",
+        max_speakers=args.max_speakers,
+        min_speakers=args.min_speakers,
+        formats=args.format,
+        pause_threshold=args.pause_threshold,
+        chunk_duration=args.chunk_duration,
+        hf_token=args.hf_token,
+    )
+
+    pipeline = TranscriptionPipeline(config)
+    try:
+        exported = pipeline.run()
+        for path in exported:
+            print(f"Saved: {path}")
+    except Exception:
+        logging.exception("Pipeline failed")
+        sys.exit(1)
+
+
+if __name__ == "__main__":
+    main()
diff --git a/transcriber/merge/__init__.py b/transcriber/merge/__init__.py
new file mode 100644
index 0000000..e69de29
diff --git a/transcriber/merge/aligner.py b/transcriber/merge/aligner.py
new file mode 100644
index 0000000..bbd9c9f
--- /dev/null
+++ b/transcriber/merge/aligner.py
@@ -0,0 +1,101 @@
+import logging
+from dataclasses import dataclass
+
+from transcriber.asr.whisper_engine import Segment
+from transcriber.diarization.pyannote_engine import SpeakerTurn
+
+logger = logging.getLogger(__name__)
+
+
+@dataclass
+class MergedSegment:
+    """Final segment with speaker label and merged text."""
+
+    speaker: str
+    start: float
+    end: float
+    text: str
+
+
+def _compute_overlap(seg_start: float, seg_end: float, turn_start: float, turn_end: float) -> float:
+    """Compute temporal overlap between two intervals in seconds."""
+    overlap_start = max(seg_start, turn_start)
+    overlap_end = min(seg_end, turn_end)
+    return max(0.0, overlap_end - overlap_start)
+
+
+def _assign_speaker(segment: Segment, speaker_turns: list[SpeakerTurn]) -> str:
+    """Assign speaker to ASR segment by maximum overlap."""
+    best_speaker = "Unknown"
+    best_overlap = 0.0
+
+    for turn in speaker_turns:
+        if turn.end < segment.start:
+            continue
+        if turn.start > segment.end:
+            break
+        overlap = _compute_overlap(segment.start, segment.end, turn.start, turn.end)
+        if overlap > best_overlap:
+            best_overlap = overlap
+            best_speaker = turn.speaker
+
+    return best_speaker
+
+
+def _normalize_speaker_labels(segments: list[MergedSegment]) -> list[MergedSegment]:
+    """Replace raw pyannote labels (SPEAKER_00) with sequential Speaker 1, Speaker 2."""
+    label_map: dict[str, str] = {}
+    counter = 1
+
+    for seg in segments:
+        if seg.speaker not in label_map and seg.speaker != "Unknown":
+            label_map[seg.speaker] = f"Speaker {counter}"
+            counter += 1
+
+    for seg in segments:
+        seg.speaker = label_map.get(seg.speaker, seg.speaker)
+
+    return segments
+
+
+def align_and_merge(
+    asr_segments: list[Segment],
+    speaker_turns: list[SpeakerTurn],
+    pause_threshold: float = 1.5,
+) -> list[MergedSegment]:
+    """Align ASR segments with speaker turns and merge adjacent same-speaker segments.
+
+    Args:
+        asr_segments: Segments from Whisper ASR.
+        speaker_turns: Speaker turns from diarization (sorted by start).
+        pause_threshold: Max pause between segments to merge (seconds).
+
+    Returns:
+        List of merged segments with speaker labels.
+    """
+    if not asr_segments:
+        return []
+
+    aligned = []
+    for seg in asr_segments:
+        speaker = _assign_speaker(seg, speaker_turns)
+        aligned.append(MergedSegment(
+            speaker=speaker,
+            start=seg.start,
+            end=seg.end,
+            text=seg.text,
+        ))
+
+    merged: list[MergedSegment] = [aligned[0]]
+    for seg in aligned[1:]:
+        prev = merged[-1]
+        gap = seg.start - prev.end
+        if seg.speaker == prev.speaker and gap <= pause_threshold:
+            prev.end = seg.end
+            prev.text = f"{prev.text} {seg.text}"
+        else:
+            merged.append(seg)
+
+    merged = _normalize_speaker_labels(merged)
+    logger.info("Alignment complete: %d ASR segments -> %d merged", len(asr_segments), len(merged))
+    return merged
diff --git a/transcriber/pipeline.py b/transcriber/pipeline.py
new file mode 100644
index 0000000..2366b26
--- /dev/null
+++ b/transcriber/pipeline.py
@@ -0,0 +1,136 @@
+import logging
+from pathlib import Path
+
+from tqdm import tqdm
+
+from transcriber.asr.whisper_engine import Segment, WhisperEngine
+from transcriber.audio.chunking import ChunkInfo, chunk_audio
+from transcriber.audio.preprocess import preprocess_audio
+from transcriber.config import TranscriberConfig
+from transcriber.diarization.pyannote_engine import DiarizationEngine
+from transcriber.export.json_writer import write_json
+from transcriber.export.txt_writer import write_txt
+from transcriber.merge.aligner import MergedSegment, align_and_merge
+
+logger = logging.getLogger(__name__)
+
+EXPORTERS = {
+    "txt": write_txt,
+    "json": write_json,
+}
+
+
+class TranscriptionPipeline:
+    """Orchestrates the full transcription pipeline: preprocess -> ASR -> diarize -> merge -> export."""
+
+    def __init__(self, config: TranscriberConfig):
+        self._config = config
+        self._asr: WhisperEngine | None = None
+        self._diarizer: DiarizationEngine | None = None
+
+    def _init_engines(self) -> None:
+        """Lazily initialize ASR and diarization engines."""
+        if self._asr is None:
+            self._asr = WhisperEngine(
+                model_name=self._config.model,
+                device=self._config.device,
+                compute_type=self._config.compute_type,
+            )
+        if self._diarizer is None:
+            self._diarizer = DiarizationEngine(
+                hf_token=self._config.hf_token,
+                device=self._config.device,
+            )
+
+    def _transcribe_chunks(self, chunks: list[ChunkInfo], progress: tqdm) -> list[Segment]:
+        """Run ASR on each chunk, adjusting timestamps by chunk offset."""
+        all_segments: list[Segment] = []
+        for chunk in chunks:
+            progress.set_description(f"ASR chunk {chunk.start_offset:.0f}s")
+            segments = self._asr.transcribe(
+                audio_path=chunk.path,
+                language=self._config.language,
+                beam_size=self._config.beam_size,
+                vad_filter=self._config.vad,
+            )
+            for seg in segments:
+                seg.start += chunk.start_offset
+                seg.end += chunk.start_offset
+                for w in seg.words:
+                    w.start += chunk.start_offset
+                    w.end += chunk.start_offset
+            all_segments.extend(segments)
+            progress.update(1)
+        return all_segments
+
+    def _export(self, segments: list[MergedSegment], stem: str) -> list[str]:
+        """Export segments to requested formats."""
+        output_dir = Path(self._config.output_dir)
+        exported = []
+        for fmt in self._config.formats:
+            exporter = EXPORTERS.get(fmt)
+            if exporter is None:
+                logger.warning("Unknown export format: %s (skipped)", fmt)
+                continue
+            out_path = str(output_dir / f"{stem}.{fmt}")
+            exporter(segments, out_path)
+            exported.append(out_path)
+            logger.info("Exported: %s", out_path)
+        return exported
+
+    def run(self) -> list[str]:
+        """Execute the full pipeline and return list of exported file paths.
+
+        Returns:
+            List of paths to exported files.
+
+        Raises:
+            FileNotFoundError: If input file does not exist.
+            RuntimeError: If any pipeline stage fails.
+        """
+        cfg = self._config
+        stem = Path(cfg.input_path).stem
+
+        total_steps = 7
+        progress = tqdm(total=total_steps, desc="Pipeline", unit="step")
+
+        progress.set_description("Preprocessing")
+        wav_path = preprocess_audio(cfg.input_path, cfg.output_dir)
+        progress.update(1)
+
+        progress.set_description("Chunking")
+        chunks = chunk_audio(wav_path, cfg.chunk_duration)
+        progress.update(1)
+
+        progress.set_description("Loading models")
+        self._init_engines()
+        progress.update(1)
+
+        asr_progress = tqdm(total=len(chunks), desc="ASR", unit="chunk", leave=False)
+        asr_segments = self._transcribe_chunks(chunks, asr_progress)
+        asr_progress.close()
+        progress.update(1)
+
+        progress.set_description("Diarizing")
+        speaker_turns = self._diarizer.diarize(
+            audio_path=wav_path,
+            min_speakers=cfg.min_speakers,
+            max_speakers=cfg.max_speakers,
+        )
+        progress.update(1)
+
+        progress.set_description("Aligning")
+        merged = align_and_merge(
+            asr_segments=asr_segments,
+            speaker_turns=speaker_turns,
+            pause_threshold=cfg.pause_threshold,
+        )
+        progress.update(1)
+
+        progress.set_description("Exporting")
+        exported = self._export(merged, stem)
+        progress.update(1)
+
+        progress.close()
+        logger.info("Pipeline complete. Files: %s", exported)
+        return exported