@lemoncat7/dsh-partner 2.8.1 → 2.8.3

package/resources/skills/video-recap-production/upstream/skills/video-voiceover/scripts/dub.py

@@ -0,0 +1,623 @@
+"""English→Chinese dubbing — the render engine behind `recap.py --edit-mode dub`.
+
+Invoked by the orchestrator (NOT run by hand). It replaces the original English speech with a
+faithful Chinese translation spoken in the ORIGINAL speaker's cloned voice
+(`mimo-v2.5-tts-voiceclone`, same MIMO_API_KEY, pure stdlib + ffmpeg, no GPU). This differs
+from recap/解说, which overlays Chinese commentary on ducked original audio.
+
+Division of labour (the same as recap): CODE does only the mechanical parts; the AGENT does all
+the judgment. So there are NO text heuristics here (no sentence-splitting, hook-dedup, or junk
+filters) — those are exactly the things an LLM does better, and trying to do them in code is
+brittle and case-by-case. Two stages around an agent-authored pause:
+
+  --stage prepare : extract audio → English ASR (timed windows) → pull one reference clip →
+                    write dub_transcript.json + dub_brief.md, then stop.
+  --stage render  : read the agent's dub_script.json ([{"start", "zh"}] on the source timeline)
+                    → clone the original voice per line → time-fit each to its slot (anchored at
+                    its start; only compress when it would overrun the next line — never a global
+                    speed-up, so the voice never finishes ahead of the picture) → full-replace
+                    the audio track → mux → dub_<name>.mp4.
+"""
+import argparse
+import base64
+import json
+import re
+import unicodedata
+import wave
+from pathlib import Path
+
+from lib import (
+    file_identity,
+    CONFIG,
+    get_video_duration,
+    log,
+    mimo_asr_api_call,
+    mimo_tts_api_call,
+    run_cmd,
+)
+
+CLONE_MODEL = "mimo-v2.5-tts-voiceclone"
+CLONE_SR = 24000  # mimo voiceclone returns 24kHz mono PCM16 wav
+DUB_DELIVERY_SR = 48000  # explicit delivery rate; loudnorm otherwise exposes its 96kHz internal rate
+ASR_MIME = "audio/wav"
+ATEMPO_CAP = 2.0  # max compression before we trim instead (atempo>2 also sounds rushed)
+DUB_FAST_SPEECH_CPS = 7.0
+DUB_TRIM_RISK_CPS = 7.0
+DUB_MIN_ASR_WINDOW_SECONDS = 0.5
+DUB_TTS_STYLE_PROMPT = "自然、清晰，保持原说话人的音色与节奏，语气平稳。"
+
+DUB_SCHEMA_VERSION = 1
+# Tolerate ~1-frame rounding in agent-estimated timings so the gate blocks only GENUINE
+# errors, not rounding noise (the old render clamped such cases via min(slot_end, nxt)).
+DUB_TIMING_EPS = 0.05
+DUB_LINT_SCHEMA = {
+    "schema_version": DUB_SCHEMA_VERSION,
+    "artifact": "dub_lint.json",
+    "fields": {
+        "verdict": "PASS|FAIL",
+        "issues": "[{severity, code, line, message, start, end}]",
+        "summary": "{lines, errors, warnings, max_chars_per_second, trim_risk_lines}",
+    },
+}
+DUB_REVIEW_SCHEMA = {
+    "schema_version": DUB_SCHEMA_VERSION,
+    "artifact": "dub_review.json",
+    "fields": {
+        "verdict": "PASS|REVISE|FAIL",
+        "checks": "{faithful_to_source, spoken_chinese, speaker_tone, timing_fit, platform_fit}",
+        "highest_return_edits": "[{line, start, issue, suggestion}]",
+        "notes": "human/agent review notes; script-level output is deterministic pre-review guidance",
+    },
+}
+DUB_ARTIFACT_SCHEMAS = {
+    "dub_transcript.json": {
+        "schema_version": DUB_SCHEMA_VERSION,
+        "shape": {"video": "path", "duration": "seconds", "windows": [{"start": 0.0, "end": 6.0, "text": "English ASR"}]},
+    },
+    "dub_script.json": {
+        "schema_version": DUB_SCHEMA_VERSION,
+        "shape": [{"start": 0.0, "end": 2.4, "zh": "中文台词"}],
+    },
+    "dub_manifest.json": {
+        "schema_version": DUB_SCHEMA_VERSION,
+        "shape": {"video": "path", "duration": "seconds", "lines": [{"start": 0.0, "end": 2.4, "zh": "中文台词", "tts_cache": "hit|miss", "fitted_wav": "path", "fitted_dur": 1.8, "room": 2.4}]},
+    },
+    "dub_lint.json": DUB_LINT_SCHEMA,
+    "dub_review.json": DUB_REVIEW_SCHEMA,
+}
+
+
+def _chars_per_second(text, room):
+    room = max(room, 0.001)
+    effective = 0
+    for ch in text:
+        if ch.isspace():
+            continue
+        cat = unicodedata.category(ch)
+        if cat.startswith(("P", "S")):
+            continue
+        effective += 1
+    return effective / room
+
+
+def _issue(severity, code, line, message, start=None, end=None):
+    item = {"severity": severity, "code": code, "line": line, "message": message}
+    if start is not None:
+        item["start"] = round(start, 3)
+    if end is not None:
+        item["end"] = round(end, 3)
+    return item
+
+
+def _normalize_dub_script(script):
+    lines = []
+    for i, item in enumerate(script):
+        if not isinstance(item, dict):
+            lines.append({"_input_index": i, "start": None, "end": None, "zh": ""})
+            continue
+        start = item.get("start")
+        end = item.get("end")
+        try:
+            start = float(start)
+        except (TypeError, ValueError):
+            start = None
+        try:
+            end = float(end) if end is not None else None
+        except (TypeError, ValueError):
+            end = None
+        lines.append({"_input_index": i, "start": start, "end": end,
+                      "zh": str(item.get("zh", "")).strip()})
+    return sorted(lines, key=lambda x: (float("inf") if x["start"] is None else x["start"], x["_input_index"]))
+
+
+def lint_dub_script(script, duration, work_dir=None):
+    """Deterministic hard gate for dub_script.json before expensive voiceclone render."""
+    duration = float(duration)
+    issues = []
+    if not isinstance(script, list):
+        # A malformed (non-list) script is a hard FAIL, not a crash: keep the "fail fast,
+        # see dub_lint.json" promise so the agent gets a clean message instead of a traceback.
+        bad = _issue("error", "script_not_a_list", None, "dub_script.json must be a JSON array of {start,end,zh}")
+        report = {
+            "schema_version": DUB_SCHEMA_VERSION,
+            "verdict": "FAIL",
+            "blocking": True,
+            "errors": [bad],
+            "issues": [bad],
+            "summary": {"lines": 0, "errors": 1, "warnings": 0, "max_chars_per_second": 0.0, "trim_risk_lines": []},
+        }
+        if work_dir is not None:
+            _write_json(Path(work_dir) / "dub_lint.json", report)
+        return report
+    lines = _normalize_dub_script(script)
+    max_cps = 0.0
+    trim_risk = []
+    for i, ln in enumerate(lines):
+        start, end, zh = ln["start"], ln["end"], ln["zh"]
+        line_no = ln["_input_index"]
+        nxt = lines[i + 1]["start"] if i + 1 < len(lines) else duration
+        if start is None:
+            issues.append(_issue("error", "missing_start", line_no, "line start must be a number"))
+            continue
+        out_of_range = start < -DUB_TIMING_EPS or start > duration + DUB_TIMING_EPS
+        if end is not None and (end <= start or end > duration + DUB_TIMING_EPS):
+            out_of_range = True
+        timing_valid = not out_of_range
+        if out_of_range:
+            issues.append(_issue("error", "time_out_of_range", line_no, "line timing must be within video duration and end must be > start", start, end))
+        if not zh:
+            issues.append(_issue("error", "empty_translation", line_no, "zh must not be empty", start, end))
+        if (i + 1 < len(lines) and end is not None and lines[i + 1]["start"] is not None
+                and end > lines[i + 1]["start"] + DUB_TIMING_EPS):
+            issues.append(_issue("error", "overlap", line_no, "line end overlaps the next line start", start, end))
+        slot_end = end if end is not None else nxt
+        if timing_valid and start is not None and slot_end is not None:
+            room = max(0.0, min(slot_end, nxt if nxt is not None else slot_end) - start)
+            cps = _chars_per_second(zh, room)
+            max_cps = max(max_cps, cps)
+            if room < 0.4:
+                issues.append(_issue("error", "room_too_short", line_no, "line has less than 0.4s room for speech", start, end))
+            elif zh and cps >= DUB_FAST_SPEECH_CPS:
+                issues.append(_issue("warning", "fast_speech", line_no, f"translation is dense ({cps:.1f} chars/s)", start, end))
+            if zh and cps >= DUB_TRIM_RISK_CPS:
+                trim_risk.append(line_no)
+                issues.append(_issue("warning", "trim_risk", line_no, "likely to be compressed hard or trimmed by render", start, end))
+    error_items = [x for x in issues if x["severity"] == "error"]
+    priority = {"empty_translation": 0, "overlap": 1, "time_out_of_range": 2}
+    error_items = sorted(error_items, key=lambda x: (priority.get(x["code"], 99), x.get("line", 0)))
+    warnings = sum(1 for x in issues if x["severity"] == "warning")
+    report = {
+        "schema_version": DUB_SCHEMA_VERSION,
+        "verdict": "FAIL" if error_items else "PASS",
+        "blocking": bool(error_items),
+        "errors": error_items,
+        "issues": issues,
+        "summary": {
+            "lines": len(lines),
+            "errors": len(error_items),
+            "warnings": warnings,
+            "max_chars_per_second": round(max_cps, 2),
+            "trim_risk_lines": trim_risk,
+        },
+    }
+    if work_dir is not None:
+        _write_json(Path(work_dir) / "dub_lint.json", report)
+    return report
+
+
+def build_dub_review(script, transcript, lint=None):
+    """Script-level review scaffold: deterministic timing/naturalness signals for agent review."""
+    duration = float(transcript["duration"])
+    lint = lint or lint_dub_script(script, duration)
+    lines = _normalize_dub_script(script) if isinstance(script, list) else []
+    edits = []
+    for issue in lint["issues"]:
+        if issue["severity"] in {"error", "warning"}:
+            edits.append({
+                "line": issue["line"],
+                "start": issue.get("start"),  # _issue only records start when the line had one
+                "issue": f"{issue['code']}: {issue['message']}",
+                "suggestion": "缩短译文、调整 start/end，或回到 transcript 核对原句边界。",
+            })
+    verdict = "FAIL" if lint["verdict"] == "FAIL" else ("REVISE" if edits else "PASS")
+    return {
+        "schema_version": DUB_SCHEMA_VERSION,
+        "verdict": verdict,
+        "checks": {
+            "faithful_to_source": "needs_agent_review",
+            "spoken_chinese": "REVISE" if any(i["code"] == "fast_speech" for i in lint["issues"]) else "PASS",
+            "speaker_tone": "needs_agent_review",
+            "timing_fit": verdict,
+            "platform_fit": "needs_agent_review",
+        },
+        "highest_return_edits": edits[:8],
+        "notes": "Deterministic script-level review; an agent/human should fill semantic fidelity and tone judgments against dub_transcript.json.",
+        "coverage": {"transcript_windows": len(transcript["windows"]), "script_lines": len(lines)},
+    }
+
+
+def _write_json(path, payload):
+    Path(path).write_text(json.dumps(payload, ensure_ascii=False, indent=2), encoding="utf-8")
+    return path
+
+
+# ── ffmpeg / wav helpers ─────────────────────────────────────────────
+
+def _ffmpeg(args):
+    """Run one ffmpeg command; a non-zero exit raises instead of leaving a silent gap."""
+    result = run_cmd(["ffmpeg", "-y", *args])
+    if result.returncode != 0:
+        raise RuntimeError(f"ffmpeg 失败 ({args[-1]}): {result.stderr.strip()[-2000:]}")
+
+
+def _ffmpeg_extract_wav(video, out_wav, sr=16000):
+    _ffmpeg(["-i", str(video), "-vn", "-ar", str(sr), "-ac", "1", str(out_wav)])
+
+
+def _cut_wav(src_wav, out_wav, start, dur, sr=16000):
+    _ffmpeg(["-i", str(src_wav), "-ss", str(start), "-t", str(dur),
+             "-ar", str(sr), "-ac", "1", str(out_wav)])
+
+
+def _atempo_chain(factor):
+    """ffmpeg atempo only accepts 0.5–2.0 per stage; chain for larger factors."""
+    factor = max(0.5, factor)
+    stages, remaining = [], factor
+    while remaining > 2.0:
+        stages.append("atempo=2.0")
+        remaining /= 2.0
+    stages.append(f"atempo={remaining:.4f}")
+    return ",".join(stages)
+
+
+def _wav_frames(path):
+    with wave.open(str(path), "rb") as w:
+        return w.getframerate(), w.getnchannels(), w.readframes(w.getnframes())
+
+
+# ── ASR (timed windows) ──────────────────────────────────────────────
+
+def _strip_reasoning_residue(text):
+    """Remove MiMo reasoning-model <think>…</think> leakage from ASR content.
+
+    Thinking-disable is not applied to -asr models (lib._prepare_api_payload), so a reasoning
+    model can leak a <think> block — full, truncated/unclosed, or a leading orphan/residual tag
+    (observed as a bare "think>" prefix) — into the transcript. Strip all shapes; the generic
+    marker strip in _run_asr then handles the remaining "<chinese>"-style tags.
+    """
+    text = re.sub(r"(?is)<think\b.*?</think\s*>", "", text)  # full <think>…</think> block
+    text = re.sub(r"(?is)<think\b.*\Z", "", text)            # unclosed/truncated <think tail
+    return re.sub(r"(?i)^\s*<?/?think\s*>\s*", "", text)     # leading orphan/residual think tag
+
+
+def _run_asr(wav_path, lang="en"):
+    # The WAV was just cut by ffmpeg (which raised on failure); an unreadable file is a bug,
+    # not "no speech". A zero-frame window (audio shorter than the picture) is skipped.
+    with wave.open(str(wav_path), "rb") as source:
+        if source.getnframes() <= 0:
+            return ""
+    b64 = base64.b64encode(Path(wav_path).read_bytes()).decode("ascii")
+    payload = {
+        "model": CONFIG["mimo_asr_model"],
+        "messages": [{"role": "user", "content": [
+            {"type": "input_audio", "input_audio": {"data": f"data:{ASR_MIME};base64,{b64}"}},
+        ]}],
+        "asr_options": {"language": lang},
+    }
+    resp = mimo_asr_api_call(payload)
+    try:
+        text = str(resp["choices"][0]["message"]["content"] or "").strip()
+    except (KeyError, IndexError, TypeError) as exc:
+        raise RuntimeError("MiMo-ASR 响应缺少 choices[0].message.content") from exc
+    return re.sub(r"<[^>]{1,20}>", "", _strip_reasoning_residue(text)).strip()  # strip reasoning + ASR markers like "<chinese>"
+
+
+def _asr_windows(audio_wav, segs_dir, duration, window):
+    """Transcribe fixed-time windows → [{start, end, text}]. Coarse timing anchors for the agent;
+    the agent does the sentence/segment judgment, not this code."""
+    windows = []
+    start, idx = 0.0, 0
+    while start < duration:
+        end = min(start + window, duration)
+        if end - start < DUB_MIN_ASR_WINDOW_SECONDS:
+            log(f"  ASR {start:.0f}-{end:.0f}s: 跳过不足 {DUB_MIN_ASR_WINDOW_SECONDS:.1f}s 的编码尾巴")
+            break
+        seg_wav = segs_dir / f"asr_{idx:03d}.wav"
+        _cut_wav(audio_wav, seg_wav, start, end - start)
+        text = _run_asr(seg_wav)
+        if text:
+            windows.append({"start": round(start, 2), "end": round(end, 2), "text": text})
+        log(f"  ASR {start:.0f}-{end:.0f}s: {len(text)} chars")
+        start, idx = end, idx + 1
+    return windows
+
+
+# ── clone TTS + time-fit + mix ───────────────────────────────────────
+
+def _clone_tts(text, ref_b64, out_wav):
+    payload = {
+        "model": CLONE_MODEL,
+        "messages": [
+            {"role": "user", "content": DUB_TTS_STYLE_PROMPT},
+            {"role": "assistant", "content": text},
+        ],
+        "audio": {"format": "wav", "voice": f"data:audio/wav;base64,{ref_b64}"},
+    }
+    resp = mimo_tts_api_call(payload)
+    data = resp["choices"][0]["message"]["audio"]["data"]
+    Path(out_wav).write_bytes(base64.b64decode(data))
+
+
+def _clone_cache_inputs(text, ref_identity):
+    return {
+        "model": CLONE_MODEL,
+        "style_prompt": DUB_TTS_STYLE_PROMPT,
+        "text": str(text),
+        "reference": ref_identity,
+    }
+
+
+def _clone_cache_meta_path(raw_wav):
+    raw_wav = Path(raw_wav)
+    return raw_wav.with_name(f"{raw_wav.name}.meta.json")
+
+
+def _usable_clone_wav(path):
+    try:
+        sample_rate, channels, frames = _wav_frames(path)
+    except (FileNotFoundError, OSError, EOFError, wave.Error):
+        return False
+    return sample_rate > 0 and channels > 0 and bool(frames)
+
+
+def _write_clone_cache_meta(raw_wav, text, ref_identity):
+    meta = {"inputs": _clone_cache_inputs(text, ref_identity)}
+    return _write_json(_clone_cache_meta_path(raw_wav), meta)
+
+
+def _ensure_clone_tts(text, ref_b64, ref_identity, raw_wav):
+    raw_wav = Path(raw_wav)
+    expected = _clone_cache_inputs(text, ref_identity)
+    meta_path = _clone_cache_meta_path(raw_wav)
+    # Missing sidecar = never synthesized; a sidecar this skill wrote but cannot parse is a
+    # bug, not a reason to silently pay for re-synthesis.
+    if meta_path.exists():
+        meta = json.loads(meta_path.read_text(encoding="utf-8"))
+        # A meta from the content-hash schema has no "inputs" and is simply a miss.
+        if meta.get("inputs") == expected and _usable_clone_wav(raw_wav):
+            return True
+
+    _clone_tts(text, ref_b64, raw_wav)
+    if not _usable_clone_wav(raw_wav):
+        raise RuntimeError(f"MiMo voiceclone returned an invalid WAV: {raw_wav}")
+    _write_clone_cache_meta(raw_wav, text, ref_identity)
+    return False
+
+
+def _time_fit(raw_wav, fitted_wav, room_seconds):
+    """Anchor-at-start fit: only compress when the dub would overrun `room_seconds` (the gap until
+    the next line). Never globally speed up; a short dub keeps the natural pause. Returns the
+    fitted duration."""
+    dur = get_video_duration(raw_wav)
+    if dur <= room_seconds + 0.05:
+        _ffmpeg(["-i", str(raw_wav), "-ar", str(CLONE_SR), "-ac", "1", str(fitted_wav)])
+        return dur
+    factor = dur / room_seconds
+    if factor <= ATEMPO_CAP:
+        _ffmpeg(["-i", str(raw_wav), "-filter:a", _atempo_chain(factor),
+                 "-ar", str(CLONE_SR), "-ac", "1", str(fitted_wav)])
+        return get_video_duration(fitted_wav)
+    _ffmpeg(["-i", str(raw_wav),
+             "-filter:a", f"{_atempo_chain(ATEMPO_CAP)},atrim=0:{room_seconds:.3f},"
+                          f"afade=t=out:st={max(0, room_seconds - 0.15):.3f}:d=0.15",
+             "-ar", str(CLONE_SR), "-ac", "1", str(fitted_wav)])
+    return get_video_duration(fitted_wav)
+
+
+def _build_dub_track(lines, duration, out_wav):
+    canvas = bytearray(int(duration * CLONE_SR) * 2)  # 16-bit mono silence
+    for ln in lines:
+        # _time_fit always writes fitted_wav at CLONE_SR mono (and raised if ffmpeg failed).
+        sr, ch, frames = _wav_frames(ln["fitted_wav"])
+        if sr != CLONE_SR or ch != 1:
+            raise RuntimeError(f"fitted WAV {ln['fitted_wav']} is {sr}Hz/{ch}ch, expected {CLONE_SR}Hz mono")
+        off = int(ln["start"] * CLONE_SR) * 2
+        end = min(off + len(frames), len(canvas))
+        canvas[off:end] = frames[: end - off]
+    with wave.open(str(out_wav), "wb") as w:
+        w.setnchannels(1)
+        w.setsampwidth(2)
+        w.setframerate(CLONE_SR)
+        w.writeframes(bytes(canvas))
+
+
+def _mux(video, dub_wav, out_video):
+    _ffmpeg(["-i", str(video), "-i", str(dub_wav),
+             "-map", "0:v:0", "-map", "1:a:0", "-c:v", "copy",
+             "-af", "loudnorm=I=-16:TP=-1.5:LRA=11", "-c:a", "aac", "-b:a", "192k",
+             "-ar", str(DUB_DELIVERY_SR),
+             "-shortest", str(out_video)])
+
+
+# ── stages ───────────────────────────────────────────────────────────
+
+def _ref_window(duration, ref_start, ref_dur):
+    start = max(0.0, min(ref_start, max(0.0, duration - 2.0)))
+    return start, max(2.0, min(ref_dur, duration - start))
+
+
+def stage_prepare(video, work, asr_window, ref_start, ref_dur):
+    work.mkdir(parents=True, exist_ok=True)
+    segs_dir = work / "dub_asr"
+    segs_dir.mkdir(exist_ok=True)
+    duration = get_video_duration(video)
+    log(f"[dub:prepare] video {duration:.1f}s")
+
+    audio_wav = work / "dub_source.wav"
+    _ffmpeg_extract_wav(video, audio_wav)
+    windows = _asr_windows(audio_wav, segs_dir, duration, asr_window)
+    if not windows:
+        raise SystemExit("[dub] no speech transcribed")
+    log(f"[dub:prepare] {len(windows)} ASR windows")
+
+    rs, rd = _ref_window(duration, ref_start, ref_dur)
+    _cut_wav(audio_wav, work / "dub_reference.wav", rs, rd)
+
+    (work / "dub_transcript.json").write_text(
+        json.dumps({"video": str(video), "duration": duration, "windows": windows},
+                   ensure_ascii=False, indent=2), encoding="utf-8")
+    (work / "dub_brief.md").write_text(_brief_md(windows, duration), encoding="utf-8")
+    print(json.dumps({"status": "dub_prepared", "windows": len(windows),
+                      "brief": str(work / "dub_brief.md")}, ensure_ascii=False))
+
+
+def _brief_md(windows, duration):
+    lines = [
+        "# 配音翻译任务（dub）",
+        "",
+        f"视频时长 {duration:.1f}s。下面是英文原声的分窗转写（时间戳偏粗，仅作锚点）。",
+        "把它**忠实翻译并配音**，写入 `dub_script.json`——核心是**和原视频节奏一致**，不是做解说、"
+        "也不是做精简版。",
+        "",
+        "要点：",
+        "1. 逐句忠实翻译，原声说什么就配什么；**不要删内容、不要合并、不要自作主张去掉开场 hook**；"
+        "原声重复，配音也跟着重复。",
+        "2. 每句放在它在原声里被说出的时间，`start`/`end` 取那句话的起止——这样配音才跟着原声节奏走"
+        "（该停顿处自然留白）。",
+        "3. 译文忠实精简，能在自己的 `start`→`end` 区间内用正常语速说完（约 5 字/秒）。",
+        "4. 只有完全被截断、没法说完整的残句，才取其中说得完整的部分。",
+        "",
+        "输出 `dub_script.json` = `[{\"start\": 起秒, \"end\": 止秒, \"zh\": \"中文台词\"}, ...]`，按 start "
+        "升序，[start,end] 是该句在原声里的时间区间，相邻句不要重叠。",
+        "",
+        "## 英文原声转写（按时间窗）",
+        "",
+        "| 起–止 | 英文 |",
+        "|---|---|",
+    ]
+    for w in windows:
+        lines.append(f"| {w['start']}–{w['end']}s | {w['text'].replace('|', '/')} |")
+    return "\n".join(lines) + "\n"
+
+
+def stage_render(video, work, ref_start, ref_dur):
+    transcript = json.loads((work / "dub_transcript.json").read_text(encoding="utf-8"))
+    duration = transcript["duration"]
+    script = json.loads((work / "dub_script.json").read_text(encoding="utf-8"))
+    # Mechanical hard gate BEFORE any voiceclone spend: empty/overlapping/out-of-range lines
+    # cannot produce a publishable dub, so fail fast instead of paying for a broken render.
+    lint = lint_dub_script(script, duration)
+    _write_json(work / "dub_lint.json", lint)
+    review = build_dub_review(script, transcript, lint)
+    _write_json(work / "dub_review.json", review)
+    if lint["verdict"] != "PASS":
+        raise SystemExit(f"[dub] dub_script.json failed lint; see {work / 'dub_lint.json'}")
+    lines = [{"start": ln["start"], "end": ln["end"], "zh": ln["zh"]}
+             for ln in _normalize_dub_script(script) if ln["zh"]]
+    if not lines:
+        raise SystemExit("[dub] dub_script.json has no lines")
+
+    ref_wav = work / "dub_reference.wav"
+    if not ref_wav.exists():
+        rs, rd = _ref_window(duration, ref_start, ref_dur)
+        _cut_wav(work / "dub_source.wav", ref_wav, rs, rd)
+    ref_identity = file_identity(ref_wav)
+    ref_b64 = base64.b64encode(ref_wav.read_bytes()).decode("ascii")
+
+    tts_dir = work / "dub_tts"
+    tts_dir.mkdir(exist_ok=True)
+    log("[dub:render] clone-TTS + time-fit per line…")
+    for i, ln in enumerate(lines):
+        nxt = lines[i + 1]["start"] if i + 1 < len(lines) else duration
+        slot_end = ln["end"] if ln["end"] is not None else nxt
+        # fit to the original utterance's span (rhythm-faithful), but never overrun the next line
+        room = max(0.4, min(slot_end, nxt) - ln["start"])
+        raw = tts_dir / f"line_{i:03d}_raw.wav"
+        fitted = tts_dir / f"line_{i:03d}.wav"
+        cache_hit = _ensure_clone_tts(ln["zh"], ref_b64, ref_identity, raw)
+        ln["tts_cache"] = "hit" if cache_hit else "miss"
+        ln["fitted_wav"] = str(fitted)
+        ln["fitted_dur"] = round(_time_fit(raw, fitted, room), 2)
+        ln["room"] = round(room, 2)
+        log(
+            f"  line {i}: {ln['start']:.1f}s fit={ln['fitted_dur']}s/room {ln['room']}s "
+            f"cache={ln['tts_cache']} «{ln['zh'][:18]}»"
+        )
+
+    dub_wav = work / "dub_track.wav"
+    _build_dub_track(lines, duration, dub_wav)
+    out_video = work / f"dub_{Path(video).stem}.mp4"
+    _mux(video, dub_wav, out_video)
+    (work / "dub_manifest.json").write_text(
+        json.dumps({"schema_version": DUB_SCHEMA_VERSION, "video": str(video), "duration": duration, "lines": lines},
+                   ensure_ascii=False, indent=2), encoding="utf-8")
+    log(f"[dub:render] done → {out_video}")
+    print(json.dumps({"status": "dubbed", "output": str(out_video), "lines": len(lines)},
+                     ensure_ascii=False))
+
+
+def stage_lint(work):
+    transcript = json.loads((work / "dub_transcript.json").read_text(encoding="utf-8"))
+    script = json.loads((work / "dub_script.json").read_text(encoding="utf-8"))
+    lint = lint_dub_script(script, transcript["duration"])
+    path = _write_json(work / "dub_lint.json", lint)
+    print(json.dumps({"status": "dub_linted", "verdict": lint["verdict"], "issues": len(lint["issues"]), "dub_lint": str(path)}, ensure_ascii=False))
+    return lint
+
+
+def stage_review(work):
+    transcript = json.loads((work / "dub_transcript.json").read_text(encoding="utf-8"))
+    script = json.loads((work / "dub_script.json").read_text(encoding="utf-8"))
+    lint = lint_dub_script(script, transcript["duration"])
+    _write_json(work / "dub_lint.json", lint)
+    review = build_dub_review(script, transcript, lint)
+    path = _write_json(work / "dub_review.json", review)
+    print(json.dumps({"status": "dub_reviewed", "verdict": review["verdict"], "edits": len(review["highest_return_edits"]), "dub_review": str(path)}, ensure_ascii=False))
+    return review
+
+
+def print_schemas():
+    print(json.dumps(DUB_ARTIFACT_SCHEMAS, ensure_ascii=False, indent=2))
+
+
+def main():
+    ap = argparse.ArgumentParser(
+        description=(
+            "English→Chinese dub workflow. Artifacts: dub_transcript.json, "
+            "agent-authored dub_script.json, dub_lint.json, dub_review.json, dub_manifest.json. "
+            "Uses MiMo ASR plus mimo-v2.5-tts-voiceclone; ordinary narration TTS config is separate."
+        )
+    )
+    ap.add_argument("--stage", choices=["prepare", "lint", "review", "render"], required=False,
+                    help="prepare ASR brief; lint/review dub_script.json; render writes lint/review before voiceclone")
+    ap.add_argument("--video", required=False, help="source video (required for prepare/render)")
+    ap.add_argument("--work-dir", required=False, help="work directory containing dub artifacts")
+    ap.add_argument("--asr-window", type=float, default=6.0)
+    ap.add_argument("--ref-start", type=float, default=2.0)
+    ap.add_argument("--ref-dur", type=float, default=10.0)
+    ap.add_argument("--print-schema", action="store_true",
+                    help="print JSON schemas for dub_transcript/script/lint/review/manifest and exit")
+    args = ap.parse_args()
+    if args.print_schema:
+        print_schemas()
+        return
+    if not args.stage:
+        ap.error("--stage is required unless --print-schema is used")
+    if not args.work_dir:
+        ap.error("--work-dir is required")
+    work = Path(args.work_dir)
+    if args.stage in {"prepare", "render"} and not args.video:
+        ap.error("--video is required for prepare/render")
+    video = Path(args.video) if args.video else None
+    if args.stage == "prepare":
+        stage_prepare(video, work, args.asr_window, args.ref_start, args.ref_dur)
+    elif args.stage == "lint":
+        stage_lint(work)
+    elif args.stage == "review":
+        stage_review(work)
+    else:
+        stage_render(video, work, args.ref_start, args.ref_dur)
+
+
+if __name__ == "__main__":
+    main()