HomePilot / backend /app /studio /alignment.py
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"""
Forced alignment for the essay-to-video pipeline (Batch 2).
Given the essay's existing narration audio and the script-mode scenes (whose
narration is the essay's own sentences, verbatim - Batch 0 guarantees this),
compute where each scene's words actually fall in the audio. Two things fall
out: real per-scene durations (cutting on sentence boundaries, not the flat
5-second default) and word-level timestamps persisted for caption
materialization in the promote-to-project bridge (Batch 4) - no re-run needed.
Two alignment tiers, best available wins:
whisperx - word-level timestamps via WhisperX (Whisper + wav2vec2).
Heavy optional dependency; imported lazily and only when
installed. Env: WHISPERX_MODEL (default large-v3),
WHISPERX_DEVICE (default cuda; falls back to cpu).
proportional - deterministic fallback: total audio duration (ffprobe, or
the wave module for WAV) distributed across scenes by word
count. No dependencies. Scene boundaries are already
sentence boundaries, so cuts stay clean - just less exact.
Alignment is strictly opt-in: no audio URL, no alignment, and the Batch 0
flat-duration behavior is untouched.
"""
from __future__ import annotations
import json
import os
import re
import shutil
import subprocess
import tempfile
import wave
from pathlib import Path
from typing import List, Optional
from pydantic import BaseModel, Field
class AlignedWord(BaseModel):
word: str
start: float
end: float
class SceneSpan(BaseModel):
scene_number: int
start_sec: float
end_sec: float
@property
def duration_sec(self) -> float:
return max(0.0, self.end_sec - self.start_sec)
class AlignmentResult(BaseModel):
ok: bool
method: str = "" # "whisperx" | "proportional"
audio_duration_sec: float = 0.0
words: List[AlignedWord] = Field(default_factory=list)
scene_spans: List[SceneSpan] = Field(default_factory=list)
message: str = ""
# ── Audio helpers ────────────────────────────────────────────────────────────
def fetch_audio(url_or_path: str, timeout: float = 60.0) -> str:
"""Return a local file path for the audio, downloading if it's a URL.
Local filesystem paths are refused unless STUDIO_ALLOW_LOCAL_AUDIO=true:
existing_audio_url is API-supplied, and quietly reading server paths from
it would be an SSRF-style hole in multi-user deployments. Single-user
installs that want to point at local files opt in explicitly."""
if not url_or_path.startswith(("http://", "https://")):
allow_local = os.getenv("STUDIO_ALLOW_LOCAL_AUDIO", "false").strip().lower() in (
"1", "true", "yes")
if not allow_local:
raise PermissionError(
"Local audio paths are disabled (set STUDIO_ALLOW_LOCAL_AUDIO=true "
"on trusted single-user installs); pass an http(s) URL instead.")
if not Path(url_or_path).is_file():
raise FileNotFoundError(url_or_path)
return url_or_path
import httpx
suffix = Path(url_or_path.split("?", 1)[0]).suffix or ".mp3"
fd, path = tempfile.mkstemp(suffix=suffix, prefix="essay-audio-")
os.close(fd)
with httpx.stream("GET", url_or_path, timeout=timeout, follow_redirects=True) as resp:
resp.raise_for_status()
with open(path, "wb") as fh:
for chunk in resp.iter_bytes():
fh.write(chunk)
return path
def audio_duration_sec(path: str) -> Optional[float]:
"""Duration via ffprobe, falling back to the wave module for WAV files."""
ffprobe = shutil.which("ffprobe")
if ffprobe:
try:
out = subprocess.run(
[ffprobe, "-v", "quiet", "-print_format", "json",
"-show_format", path],
capture_output=True, check=True, timeout=30,
)
dur = float(json.loads(out.stdout).get("format", {}).get("duration", 0))
if dur > 0:
return dur
except Exception:
pass
if path.lower().endswith(".wav"):
try:
with wave.open(path, "rb") as wf:
return wf.getnframes() / float(wf.getframerate())
except Exception:
pass
return None
# ── WhisperX tier ────────────────────────────────────────────────────────────
def whisperx_available() -> bool:
try:
import whisperx # noqa: F401
return True
except Exception:
return False
def _whisperx_words(audio_path: str) -> List[AlignedWord]:
"""Transcribe + align, returning word-level timestamps."""
import whisperx
device = os.getenv("WHISPERX_DEVICE", "cuda").strip() or "cuda"
model_name = os.getenv("WHISPERX_MODEL", "large-v3").strip() or "large-v3"
try:
model = whisperx.load_model(model_name, device)
except Exception:
device = "cpu"
model = whisperx.load_model(model_name, device, compute_type="int8")
audio = whisperx.load_audio(audio_path)
result = model.transcribe(audio)
align_model, align_meta = whisperx.load_align_model(
language_code=result["language"], device=device)
aligned = whisperx.align(result["segments"], align_model, align_meta,
audio, device)
words: List[AlignedWord] = []
for seg in aligned.get("word_segments", []):
w = str(seg.get("word", "")).strip()
if w and seg.get("start") is not None and seg.get("end") is not None:
words.append(AlignedWord(word=w, start=float(seg["start"]),
end=float(seg["end"])))
return words
# ── Scene span mapping ───────────────────────────────────────────────────────
_WORD_RE = re.compile(r"[\w']+")
def _word_count(text: str) -> int:
return len(_WORD_RE.findall(text))
def _spans_from_words(scenes: List[dict], words: List[AlignedWord],
audio_duration: float) -> List[SceneSpan]:
"""
Map scenes onto the aligned word list by cumulative word position.
The narration is the same text the audio narrates (Batch 0's verbatim
guarantee), but the transcript may differ slightly (numbers, hyphens),
so exact token matching is brittle. Cumulative-proportional indexing
into the *aligned* word list keeps boundaries on real word timestamps
while tolerating small transcription drift.
"""
counts = [max(1, _word_count(s.get("narration", ""))) for s in scenes]
total = sum(counts)
n_words = len(words)
spans: List[SceneSpan] = []
cursor = 0
consumed = 0
for i, scene in enumerate(scenes):
consumed += counts[i]
end_idx = min(n_words - 1, round(consumed / total * n_words) - 1)
start_sec = words[cursor].start if cursor < n_words else audio_duration
end_sec = words[end_idx].end if end_idx >= cursor else start_sec
if i == len(scenes) - 1:
end_sec = max(end_sec, audio_duration)
spans.append(SceneSpan(scene_number=scene.get("scene_number", i + 1),
start_sec=round(start_sec, 3),
end_sec=round(end_sec, 3)))
cursor = min(n_words - 1, end_idx + 1)
return spans
def _spans_proportional(scenes: List[dict], audio_duration: float) -> List[SceneSpan]:
counts = [max(1, _word_count(s.get("narration", ""))) for s in scenes]
total = sum(counts)
spans: List[SceneSpan] = []
t = 0.0
for i, scene in enumerate(scenes):
dur = audio_duration * counts[i] / total
end = audio_duration if i == len(scenes) - 1 else t + dur
spans.append(SceneSpan(scene_number=scene.get("scene_number", i + 1),
start_sec=round(t, 3), end_sec=round(end, 3)))
t = end
return spans
# ── Entry points ─────────────────────────────────────────────────────────────
def align_scenes(audio_path: str, scenes: List[dict]) -> AlignmentResult:
"""Align scenes against a local audio file. Never raises for missing
optional dependencies - degrades to the proportional tier."""
duration = audio_duration_sec(audio_path)
if not duration or duration <= 0:
return AlignmentResult(ok=False, message="Could not determine audio duration "
"(is ffprobe installed?)")
if whisperx_available():
try:
words = _whisperx_words(audio_path)
if words:
return AlignmentResult(
ok=True, method="whisperx", audio_duration_sec=duration,
words=words, scene_spans=_spans_from_words(scenes, words, duration),
)
except Exception as e:
print(f"[Alignment] whisperx failed ({e}); falling back to proportional")
return AlignmentResult(
ok=True, method="proportional", audio_duration_sec=duration,
scene_spans=_spans_proportional(scenes, duration),
)
def align_from_url(audio_url: str, scenes: List[dict]) -> AlignmentResult:
"""Fetch audio (if remote) and align. Cleans up its own temp download."""
path = fetch_audio(audio_url)
is_temp = path != audio_url
try:
return align_scenes(path, scenes)
finally:
if is_temp:
try:
os.unlink(path)
except OSError:
pass
class CaptionCue(BaseModel):
start_sec: float
end_sec: float
text: str
def caption_cues(result: AlignmentResult, scenes: List[dict],
max_chars: int = 42) -> List[CaptionCue]:
"""
Build caption cues from a persisted AlignmentResult - the data source
for CaptionSegment rows in promote-to-project (Batch 4), with no
re-run of alignment.
whisperx tier: group aligned words into <= max_chars cues on real word
timestamps. proportional tier: chunk each scene's narration and spread
the chunks evenly across the scene's span.
"""
cues: List[CaptionCue] = []
if result.words:
line: List[str] = []
start = result.words[0].start
for w in result.words:
probe = " ".join(line + [w.word])
if line and len(probe) > max_chars:
cues.append(CaptionCue(start_sec=round(start, 3),
end_sec=round(w.start, 3),
text=" ".join(line)))
line, start = [w.word], w.start
else:
line.append(w.word)
if line:
cues.append(CaptionCue(start_sec=round(start, 3),
end_sec=round(result.words[-1].end, 3),
text=" ".join(line)))
return cues
by_number = {sp.scene_number: sp for sp in result.scene_spans}
for i, scene in enumerate(scenes):
span = by_number.get(scene.get("scene_number", i + 1))
text = (scene.get("narration") or "").strip()
if not span or not text:
continue
words = text.split()
chunks: List[str] = []
line: List[str] = []
for w in words:
if line and len(" ".join(line + [w])) > max_chars:
chunks.append(" ".join(line))
line = [w]
else:
line.append(w)
if line:
chunks.append(" ".join(line))
step = span.duration_sec / max(1, len(chunks))
for j, chunk in enumerate(chunks):
cues.append(CaptionCue(
start_sec=round(span.start_sec + j * step, 3),
end_sec=round(span.start_sec + (j + 1) * step, 3),
text=chunk))
return cues
def apply_spans_to_scenes(scenes: List[dict], result: AlignmentResult) -> int:
"""Write real durations (and audio offsets, for Batch 4 caption/audio
slicing) into outline scene dicts. Returns how many scenes were updated."""
by_number = {sp.scene_number: sp for sp in result.scene_spans}
updated = 0
for i, scene in enumerate(scenes):
span = by_number.get(scene.get("scene_number", i + 1))
if not span or span.duration_sec <= 0:
continue
scene["duration_sec"] = round(span.duration_sec, 3)
scene["audio_start_sec"] = span.start_sec
scene["audio_end_sec"] = span.end_sec
updated += 1
return updated