diff --git a/kb_pipeline/processors/__init__.py b/kb_pipeline/processors/__init__.py
new file mode 100644
index 0000000..be2f254
--- /dev/null
+++ b/kb_pipeline/processors/__init__.py
@@ -0,0 +1,9 @@
+"""Processors module for KB Pipeline
+
+This module contains processors for various content transformation tasks:
+- ASRProcessor: Audio-to-text conversion (subtitle download + Whisper.cpp)
+"""
+
+from .asr import ASRProcessor
+
+__all__ = ["ASRProcessor"]
diff --git a/kb_pipeline/processors/asr.py b/kb_pipeline/processors/asr.py
new file mode 100644
index 0000000..824a94b
--- /dev/null
+++ b/kb_pipeline/processors/asr.py
@@ -0,0 +1,575 @@
+"""ASRProcessor - Audio-to-text conversion
+
+Unified processor for B站官方字幕下载 and Whisper.cpp 语音识别.
+Follows the pipeline pattern established in Phase 1/2.
+
+Strategy:
+1. Try official subtitle first (highest quality, no compute cost)
+2. Fall back to Whisper.cpp if no subtitle available
+3. Auto-enable speaker diarization if HF_TOKEN is configured
+"""
+
+from __future__ import annotations
+
+import asyncio
+import json
+import logging
+import os
+import subprocess
+from pathlib import Path
+from typing import Optional, Dict, Any, List
+
+import aiohttp
+
+from ..config import get_settings
+from ..schemas import ASRInput, ASROutput
+from ..state import WorkflowStateManager
+
+logger = logging.getLogger(__name__)
+
+
+class WhisperNotFoundError(RuntimeError):
+    """Whisper.cpp 路径未找到"""
+    pass
+
+
+def get_whisper_cpp_path() -> str:
+    """获取 whisper.cpp 安装路径
+
+    检查环境变量 WHISPER_CPP_PATH 或默认路径 ~/whisper.cpp
+
+    Returns:
+        whisper.cpp 目录路径
+
+    Raises:
+        WhisperNotFoundError: 如果路径不存在
+    """
+    # 检查环境变量
+    path = os.environ.get('WHISPER_CPP_PATH')
+    if path and os.path.exists(path):
+        return os.path.expanduser(path)
+
+    # 检查默认路径
+    default = os.path.expanduser('~/whisper.cpp')
+    if os.path.exists(default):
+        return default
+
+    raise WhisperNotFoundError(
+        "未找到 whisper.cpp。请设置 WHISPER_CPP_PATH 环境变量或安装到 ~/whisper.cpp"
+    )
+
+
+def convert_to_wav(input_path: str, output_path: Optional[str] = None) -> str:
+    """使用 ffmpeg 将音频/视频转换为 16kHz WAV 格式
+
+    Args:
+        input_path: 输入文件路径
+        output_path: 输出文件路径（可选，默认为 input_path_audio.wav）
+
+    Returns:
+        输出 WAV 文件路径
+    """
+    if output_path is None:
+        base = os.path.splitext(input_path)[0]
+        output_path = f"{base}_audio.wav"
+
+    cmd = [
+        'ffmpeg', '-y', '-i', input_path,
+        '-ar', '16000',  # 16kHz 采样率
+        '-ac', '1',      # 单声道
+        '-c:a', 'pcm_s16le',  # 16-bit PCM
+        output_path
+    ]
+
+    subprocess.run(cmd, capture_output=True, check=True)
+    return output_path
+
+
+class ASRProcessor:
+    """ASR 处理器 - 统一字幕下载和语音识别
+
+    处理流程：
+    1. 如果是 B站视频，尝试下载官方字幕
+    2. 无官方字幕则调用 Whisper.cpp
+    3. HF_TOKEN 配置时自动启用说话者分离
+    4. 记录处理状态到 workflow_state.json
+
+    Attributes:
+        STEP_NAME: 步骤名称，用于状态记录
+        settings: KB 配置
+        state_manager: 状态管理器（可选）
+    """
+
+    STEP_NAME = "asr"
+
+    def __init__(self, state_manager: Optional[WorkflowStateManager] = None):
+        """初始化 ASR 处理器
+
+        Args:
+            state_manager: 状态管理器（可选，用于记录处理状态）
+        """
+        self.settings = get_settings()
+        self.state_manager = state_manager
+
+    async def process(self, input_data: ASRInput) -> ASROutput:
+        """执行 ASR 处理
+
+        策略：
+        1. 先尝试官方字幕（仅 B站）
+        2. 无官方字幕则调用 Whisper
+        3. 记录状态
+
+        Args:
+            input_data: ASR 输入数据
+
+        Returns:
+            ASR 处理结果
+
+        Raises:
+            Exception: 处理失败时抛出异常
+        """
+        result = None
+
+        try:
+            # Step 1: 尝试官方字幕（仅 B站）
+            if input_data.platform == "bilibili":
+                result = await self._try_official_subtitle(input_data)
+
+            # Step 2: 无官方字幕则调用 Whisper
+            if result is None:
+                result = await self._run_whisper(input_data)
+
+            # Step 3: 记录成功状态
+            if self.state_manager:
+                state = self.state_manager.load()
+                if state:
+                    summary = f"tool={result.tool}, segments={len(result.segments)}"
+                    if result.speakers:
+                        summary += f", speakers={len(result.speakers)}"
+                    state.mark_step_completed(self.STEP_NAME, summary)
+                    self.state_manager.save(state)
+
+            return result
+
+        except Exception as e:
+            # 记录失败状态（只在所有重试都失败后）
+            if self.state_manager:
+                state = self.state_manager.load()
+                if state:
+                    state.mark_step_failed(self.STEP_NAME, str(e))
+                    self.state_manager.save(state)
+            raise
+
+    async def _try_official_subtitle(self, input_data: ASRInput) -> Optional[ASROutput]:
+        """尝试下载 B站官方字幕
+
+        Args:
+            input_data: ASR 输入数据
+
+        Returns:
+            如果找到字幕返回 ASROutput，否则返回 None
+
+        Note:
+            无字幕返回 None（不是抛异常），让流程继续到 Whisper
+        """
+        video_path = Path(input_data.video_path)
+        output_dir = video_path.parent
+        base_name = video_path.stem
+
+        # 输出文件路径
+        txt_path = output_dir / f"{base_name}.txt"
+        srt_path = output_dir / f"{base_name}.srt"
+        json_path = output_dir / f"{base_name}.json"
+
+        async with aiohttp.ClientSession() as session:
+            try:
+                # Step 1: 获取 CID
+                bvid = input_data.content_id
+                pagelist_url = f"https://api.bilibili.com/x/player/pagelist?bvid={bvid}"
+                async with session.get(pagelist_url) as resp:
+                    pagelist_data = await resp.json()
+
+                pages = pagelist_data.get('data', [])
+                if not pages:
+                    logger.info(f"No pagelist data for {bvid}")
+                    return None
+
+                cid = pages[0].get('cid')
+                if not cid:
+                    logger.info(f"No cid found for {bvid}")
+                    return None
+
+                # Step 2: 获取字幕列表
+                player_url = f"https://api.bilibili.com/x/player/v2?bvid={bvid}&cid={cid}"
+                async with session.get(player_url) as resp:
+                    player_data = await resp.json()
+
+                subtitle_info = player_data.get('data', {}).get('subtitle', {})
+                subtitles = subtitle_info.get('list', [])
+
+                if not subtitles:
+                    logger.info(f"No official subtitle for {bvid}")
+                    return None
+
+                # Step 3: 下载第一个字幕（优先 AI 生成的中文字幕）
+                subtitle_url = None
+                for sub in subtitles:
+                    if sub.get('lan', '').startswith('zh'):
+                        subtitle_url = sub.get('subtitle_url')
+                        break
+
+                if not subtitle_url and subtitles:
+                    subtitle_url = subtitles[0].get('subtitle_url')
+
+                if not subtitle_url:
+                    return None
+
+                # 补全 URL
+                if subtitle_url.startswith('//'):
+                    subtitle_url = f"https:{subtitle_url}"
+
+                async with session.get(subtitle_url) as resp:
+                    subtitle_content = await resp.json()
+
+                # Step 4: 转换为输出格式
+                segments = []
+                text_parts = []
+                body = subtitle_content.get('body', [])
+
+                for i, item in enumerate(body):
+                    start = item.get('from', 0)
+                    end = item.get('to', 0)
+                    content = item.get('content', '')
+
+                    segments.append({
+                        'id': i,
+                        'start': start,
+                        'end': end,
+                        'text': content
+                    })
+                    text_parts.append(content)
+
+                full_text = ' '.join(text_parts)
+                duration = segments[-1]['end'] if segments else 0
+
+                # Step 5: 写入文件
+                # TXT 格式
+                txt_path.write_text(full_text, encoding='utf-8')
+
+                # SRT 格式
+                srt_content = self._convert_to_srt(segments)
+                srt_path.write_text(srt_content, encoding='utf-8')
+
+                # JSON 格式
+                json_data = {
+                    'success': True,
+                    'tool': 'subtitle',
+                    'segments': segments,
+                    'text': full_text,
+                    'duration_seconds': duration
+                }
+                json_path.write_text(json.dumps(json_data, ensure_ascii=False, indent=2), encoding='utf-8')
+
+                return ASROutput(
+                    success=True,
+                    tool="subtitle",
+                    has_subtitle=True,
+                    output_files={
+                        'txt': str(txt_path),
+                        'srt': str(srt_path),
+                        'json': str(json_path)
+                    },
+                    segments=segments,
+                    text=full_text,
+                    duration_seconds=duration
+                )
+
+            except aiohttp.ClientError as e:
+                logger.warning(f"Failed to fetch subtitle: {e}")
+                return None
+
+    def _convert_to_srt(self, segments: List[Dict]) -> str:
+        """将字幕分段转换为 SRT 格式
+
+        Args:
+            segments: 字幕分段列表，每个包含 start, end, text
+
+        Returns:
+            SRT 格式字符串
+        """
+        lines = []
+        for i, seg in enumerate(segments, 1):
+            start_time = self._seconds_to_srt_time(seg['start'])
+            end_time = self._seconds_to_srt_time(seg['end'])
+            lines.append(f"{i}")
+            lines.append(f"{start_time} --> {end_time}")
+            lines.append(seg['text'])
+            lines.append("")  # 空行
+
+        return '\n'.join(lines)
+
+    def _seconds_to_srt_time(self, seconds: float) -> str:
+        """将秒数转换为 SRT 时间格式 (HH:MM:SS,mmm)"""
+        hours = int(seconds // 3600)
+        minutes = int((seconds % 3600) // 60)
+        secs = int(seconds % 60)
+        millis = int((seconds % 1) * 1000)
+        return f"{hours:02d}:{minutes:02d}:{secs:02d},{millis:03d}"
+
+    async def _run_whisper(self, input_data: ASRInput) -> ASROutput:
+        """执行 Whisper 语音识别
+
+        根据 HF_TOKEN 配置自动选择普通模式或说话者分离模式。
+
+        Args:
+            input_data: ASR 输入数据
+
+        Returns:
+            ASR 处理结果
+        """
+        hf_token = self.settings.hf_token.get_secret_value()
+        if hf_token:
+            return await self._run_whisper_with_diarization(input_data)
+        else:
+            return await self._run_whisper_basic(input_data)
+
+    async def _run_whisper_basic(self, input_data: ASRInput) -> ASROutput:
+        """执行普通 Whisper.cpp 语音识别
+
+        Args:
+            input_data: ASR 输入数据
+
+        Returns:
+            ASR 处理结果
+
+        Raises:
+            RuntimeError: Whisper.cpp 路径不存在或执行失败
+        """
+        video_path = Path(input_data.video_path)
+        output_dir = video_path.parent
+        base_name = video_path.stem
+
+        # 获取 Whisper.cpp 路径
+        try:
+            whisper_path = get_whisper_cpp_path()
+        except WhisperNotFoundError as e:
+            raise RuntimeError(str(e))
+
+        # 模型路径
+        model_path = os.path.join(whisper_path, 'models', 'ggml-base.bin')
+        if not os.path.exists(model_path):
+            raise RuntimeError(f"Whisper model not found: {model_path}")
+
+        # 转换音频格式
+        audio_path = convert_to_wav(str(video_path))
+        temp_audio_created = audio_path != str(video_path)
+
+        try:
+            # 构建 Whisper 命令
+            output_base = os.path.join(output_dir, base_name)
+            cmd = [
+                os.path.join(whisper_path, 'whisper-cli'),
+                '-m', model_path,
+                '-f', audio_path,
+                '-of', output_base,
+                '-l', 'zh',
+                '--output-txt', '--output-srt', '--output-json',
+            ]
+
+            # 执行 Whisper
+            result = subprocess.run(cmd, capture_output=True, text=True, timeout=3600)
+
+            if result.returncode != 0:
+                raise RuntimeError(f"Whisper failed: {result.stderr}")
+
+            # 解析输出
+            return self._parse_whisper_output(output_dir, base_name)
+
+        finally:
+            # 清理临时音频文件
+            if temp_audio_created and os.path.exists(audio_path):
+                try:
+                    os.remove(audio_path)
+                except Exception:
+                    pass  # 静默失败，不阻塞主流程
+
+    def _parse_whisper_output(self, output_dir: Path, base_name: str) -> ASROutput:
+        """解析 Whisper 输出文件
+
+        Args:
+            output_dir: 输出目录
+            base_name: 文件基础名
+
+        Returns:
+            ASR 处理结果
+        """
+        txt_path = output_dir / f"{base_name}.txt"
+        srt_path = output_dir / f"{base_name}.srt"
+        json_path = output_dir / f"{base_name}.json"
+
+        # 读取 JSON 输出
+        segments = []
+        text = ""
+        duration = 0.0
+
+        if json_path.exists():
+            with open(json_path, 'r', encoding='utf-8') as f:
+                data = json.load(f)
+                # Whisper.cpp JSON 格式
+                if 'transcription' in data:
+                    for i, item in enumerate(data['transcription']):
+                        segments.append({
+                            'id': i,
+                            'start': item.get('offsets', [0, 0])[0] / 1000.0,
+                            'end': item.get('offsets', [0, 0])[1] / 1000.0,
+                            'text': item.get('text', '')
+                        })
+                text = ' '.join(s['text'] for s in segments)
+                if segments:
+                    duration = segments[-1]['end']
+
+        # 读取文本
+        if txt_path.exists():
+            text = txt_path.read_text(encoding='utf-8').strip()
+
+        return ASROutput(
+            success=True,
+            tool="whisper",
+            has_subtitle=False,
+            output_files={
+                'txt': str(txt_path),
+                'srt': str(srt_path),
+                'json': str(json_path)
+            },
+            segments=segments,
+            text=text,
+            duration_seconds=duration
+        )
+
+    async def _run_whisper_with_diarization(self, input_data: ASRInput) -> ASROutput:
+        """执行带说话者分离的 Whisper
+
+        Args:
+            input_data: ASR 输入数据
+
+        Returns:
+            ASR 处理结果，包含说话者信息
+        """
+        # 先运行普通 Whisper
+        basic_result = await self._run_whisper_basic(input_data)
+
+        try:
+            # 运行说话者分离
+            diarization_result = run_diarization(input_data.video_path)
+
+            # 合并结果
+            merged_segments = assign_speakers_to_segments(
+                basic_result.segments,
+                diarization_result
+            )
+
+            # 提取说话者列表
+            speakers = list(set(
+                seg.get('speaker') for seg in merged_segments if seg.get('speaker')
+            ))
+
+            return ASROutput(
+                success=True,
+                tool="whisper-diarization",
+                has_subtitle=False,
+                output_files=basic_result.output_files,
+                segments=merged_segments,
+                text=basic_result.text,
+                duration_seconds=basic_result.duration_seconds,
+                speakers=sorted(speakers)
+            )
+
+        except ImportError:
+            logger.warning("pyannote.audio not available, falling back to basic whisper")
+            return basic_result
+        except Exception as e:
+            logger.warning(f"Diarization failed: {e}, falling back to basic whisper")
+            return basic_result
+
+
+def run_diarization(audio_path: str, num_speakers: Optional[int] = None) -> Dict[str, List[Dict]]:
+    """运行 pyannote.audio 说话者分离
+
+    Args:
+        audio_path: 音频文件路径
+        num_speakers: 预期说话者数量（可选）
+
+    Returns:
+        说话者时间线字典 {speaker_name: [{start, end}, ...]}
+
+    Raises:
+        ImportError: 如果 pyannote.audio 未安装
+        RuntimeError: 如果 HF_TOKEN 未配置
+    """
+    hf_token = os.environ.get('HF_TOKEN')
+    if not hf_token:
+        raise RuntimeError("需要设置 HF_TOKEN 环境变量以使用说话者分离功能")
+
+    from huggingface_hub import login
+    from pyannote.audio import Pipeline
+    import torch
+
+    login(token=hf_token)
+
+    pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization-3.1")
+
+    device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
+    pipeline.to(device)
+
+    diarization = pipeline(audio_path, num_speakers=num_speakers)
+
+    # 解析结果
+    speakers = {}
+    for turn, _, speaker in diarization.itertracks(yield_label=True):
+        if speaker not in speakers:
+            speakers[speaker] = []
+        speakers[speaker].append({
+            "start": round(turn.start, 2),
+            "end": round(turn.end, 2)
+        })
+
+    return speakers
+
+
+def assign_speakers_to_segments(
+    segments: List[Dict],
+    diarization: Dict[str, List[Dict]]
+) -> List[Dict]:
+    """为每个段落分配说话者
+
+    根据时间戳匹配说话者分离结果到 Whisper 段落。
+
+    Args:
+        segments: Whisper 段落列表
+        diarization: 说话者分离结果
+
+    Returns:
+        带有 speaker 字段的段落列表
+    """
+    result = []
+
+    for seg in segments:
+        seg_start = seg.get('start', 0)
+        seg_end = seg.get('end', 0)
+        seg_mid = (seg_start + seg_end) / 2
+
+        # 找到包含中点时间的说话者
+        assigned_speaker = None
+        for speaker, turns in diarization.items():
+            for turn in turns:
+                if turn['start'] <= seg_mid <= turn['end']:
+                    assigned_speaker = speaker
+                    break
+            if assigned_speaker:
+                break
+
+        new_seg = dict(seg)
+        new_seg['speaker'] = assigned_speaker
+        result.append(new_seg)
+
+    return result