Files
VoiceStudio/backend/api/routers/capture.py
T
Palash DebnathandClaude Opus 4.7 a1ef66c321 Stability pass: DB leaks, App.jsx hooks refactor, desktop bootstrap (#49)
* fix: eliminate DB connection leaks, race conditions, and deprecated asyncio API

## DB Connection Leaks (P0)
- Convert 38 raw get_db() calls to db_conn() context manager across 14 router files
- Connections are now guaranteed to close even when exceptions are raised
- profiles.py create_profile: clean up orphaned audio file if DB insert fails
- profiles.py lock_profile: consolidate 3 separate conn.close() error paths

## Race Condition (P1)
- Add _dub_jobs_lock (threading.Lock) to protect _dub_jobs dict in dub_pipeline.py
- get_job/put_job now thread-safe for concurrent dub sessions

## asyncio Deprecation (P2)
- Replace 23 asyncio.get_event_loop() calls with asyncio.get_running_loop()
- Prevents DeprecationWarning on Python 3.12+ and future breakage on 3.14

## Quick Fixes
- gallery.py preview_voice: remove filesystem path from error response (P2)
- dub_pipeline.py parse_vtt_segments: remove redundant `import re` inside loop (P3)
- gallery.py _init_gallery_db: use db_conn() context manager (P2)

* refactor: extract hooks, centralize isTauri, add pytest-cov

## Frontend
- Extract useTTS hook (150 LOC) — TTS generation, streaming, audio ingestion
- Extract useProfiles hook (219 LOC) — voice profile CRUD, lock/unlock, preview
- Centralize isTauri detection: dialog.js, VoiceGallery.jsx, Settings.jsx
  now import from utils/media.js instead of 4 different detection patterns

## Backend
- Add pytest-cov to dev dependencies
- Baseline coverage: 39% across backend/ (214 tests pass)
- Add .coverage to .gitignore

* feat: add Vitest + checkJs, extract useDubWorkflow + useAppData hooks

## Frontend Testing (new)
- Set up Vitest with jsdom environment + @testing-library/react
- 11 tests: utils (isTauri, formatTime, constants) + Zustand store (mode, text, dubStep, pill)
- Scripts: 'test' (vitest run), 'test:watch' (vitest), 'test:legacy' (node runner)

## App.jsx Decomposition (continued)
- Extract useDubWorkflow hook (387 LOC) — upload, ingest, transcribe SSE,
  translate, generate SSE, abort, stop, cleanup
- Extract useAppData hook (181 LOC) — data loading, localStorage persistence,
  WebSocket real-time updates, model-status pill management

## TypeScript checkJs
- Enable checkJs: true in tsconfig.json for IDE-level type checking
- 947 existing errors (informational, not blocking builds)
- noImplicitAny remains false to avoid blocking

* ci: add Vitest step, fix useProfiles duplicate state

## CI
- Add 'Run Vitest (frontend)' step — runs 11 unit tests
- Override --checkJs false in CI typecheck to avoid 947 pre-existing errors
- Rename legacy test step for clarity

## Hooks
- Fix useProfiles to accept loadProfiles from parent (useAppData)
  instead of managing its own duplicate profiles array

* refactor: wire hooks into App.jsx — 2067 → 1129 LOC (-45%)

App.jsx now delegates to extracted hooks instead of inline logic:
- useAppData: data loading, localStorage, WebSocket, model pill
- useProfiles: voice profile CRUD, lock/unlock, preview
- useTTS: generation, streaming, audio ingestion
- useDubWorkflow: upload, transcribe SSE, translate, generate SSE

988 lines removed. All handler logic lives in focused,
independently testable hooks. Store selectors and render
JSX stay in App.jsx as the shell.

Verified: vite build clean, 11 frontend + 214 backend tests pass.

* feat: show real-time percentage on model loading pill

Backend: register hf_progress listener during _load_model_sync()
so download/weight-loading tqdm events update _loading_detail with
a progress percentage (0-99%). get_model_status() now includes a
'progress' field that the frontend polls.

Frontend: useAppData reads msQuery.data.progress and calls
setPillProgress() — the FloatingPill already renders the percentage
text and progress bar width from this value.

* fix: prevent FileNotFoundError in desktop bundle during model init

transformers >=4.52 calls _can_set_experts_implementation() and
_can_set_attn_implementation() during PreTrainedModel.__init__,
which open the class source file via open(class_file). In a Tauri
desktop bundle, module.__file__ points to a path that doesn't
exist on disk, causing:

  FileNotFoundError: .../omnivoice/models/omnivoice.py

Override both classmethods on OmniVoice to return static values
without filesystem access. OmniVoice doesn't use MoE experts
(return False), but does support flex/flash attn (return True).

* fix: sync source dirs on every bootstrap, not just first run

The Tauri bootstrap previously only copied omnivoice/ and backend/
to Application Support on the first run. Subsequent app updates
kept using stale source files, preventing bug fixes from landing.

Now ensure_venv_ready() always syncs both directories from the
bundle resources before returning, even when the venv is healthy.
This fixes the FileNotFoundError crash where the old omnivoice.py
lacked the _can_set_experts_implementation override.

* ui: premium setup wizard polish

- Primary button: solid gradient fill with hover glow + lift + press
- Stepper nav: connected pills with glow ring on active step
- Welcome cards: glassmorphism with stagger-in animations, lucide icons,
  left-border accent strip, hover translate
- Preflight panel: colored icon pill backgrounds, stagger-slide entrance
- Step transitions: fade+slide animation via keyed wrapper
- Footnote: shortened paths (~/ notation), Reveal in Finder button
- Recommendation banner: gradient background with accent glow
- Compact spacing throughout for denser, professional layout

* fix: kill zombie backend on clean+retry bootstrap

When clean_and_retry_bootstrap removes the project dir, any old
uvicorn process still running from the deleted paths remains alive
on port 3900. The subsequent retry_bootstrap sees the port is
healthy and attaches to the zombie instead of re-bootstrapping.

Now explicitly kill any process on the backend port after cleaning,
before calling retry_bootstrap.

* feat: integrate speaker clones into dubbing interface, sanitize system environment variables for subprocesses, and improve FFMPEG binary path resolution.

* fix: restore docker compose default + drop dead setSeed call

- deploy/docker-compose.yml: remove profiles: ["cpu"] from the default
  service so `docker compose up` matches the comment on line 5. With the
  profile present, no service auto-started.

- frontend/src/App.jsx: drop the setSeed call in restoreHistory. The
  selector was never reintroduced after the App.jsx hooks split, and
  there is no seed state in the store — seeds are generated fresh per
  call in useTTS and only read from history items for display.

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>

* fix: address CodeRabbit review — async detection, dub stream, bootstrap fail-fast

- backend/services/tts_backend.py: invert async-context detection in
  _ensure_loaded. The previous code unconditionally caught its own
  diagnostic RuntimeError and then called asyncio.run() inside a
  running loop, masking the intended error message.

- frontend/src/hooks/useDubWorkflow.js: require a terminal `done` event
  before reporting dub success. Without this, a dropped stream after
  partial progress would flip the UI to `done`, refresh history, and
  play the completion ping as if generation finished.

- frontend/src/hooks/useDubWorkflow.js: restore the previous step when
  tasksCancel() fails. The UI was getting stuck in `stopping` forever
  on cancel errors.

- frontend/src-tauri/src/bootstrap.rs: fail-fast when source sync fails
  after the existing directory has already been removed. The previous
  warn-and-continue path could leave the install with no backend/ or
  omnivoice/ sources and defer the failure to backend startup with a
  cryptic error.

- backend/api/routers/generation.py: add `from e` to the ValueError →
  HTTPException re-raise (Ruff B904).

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>

* fix: preserve % suffix in TTS generation timer

The 100ms timer in useTTS was rewriting generationTime to a plain
elapsed-seconds string, which immediately wiped the "(xx%)" download
suffix written on the next iteration of the response-body loop. The
real-time percentage was flickering on/off as a result.

Read the previous value inside the setter and reattach any existing
percent suffix.

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>

---------

Co-authored-by: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
2026-05-12 21:49:04 +05:30

127 lines
4.4 KiB
Python

"""
Standalone transcription endpoint for the Capture / Dictation feature.
Unlike /dub/transcribe/{job_id}, this endpoint is job-free — callers POST
raw audio bytes and get back transcribed text immediately. Used by:
• The frontend "Capture" (global hotkey dictation) mode
• The MCP server's future `transcribe_audio` tool
• CLI consumers that just want speech-to-text
The ASR engine is whatever `get_active_asr_backend()` returns — WhisperX
by default, or MLX Whisper on Apple Silicon when configured.
"""
from __future__ import annotations
import io
import logging
import os
import tempfile
import time
from fastapi import APIRouter, File, Form, HTTPException, UploadFile
from typing import Optional
router = APIRouter()
logger = logging.getLogger("omnivoice.capture")
@router.post("/transcribe")
async def transcribe_audio(
audio: UploadFile = File(...),
language: Optional[str] = Form(None),
model: Optional[str] = Form(None),
mode: Optional[str] = Form(None),
):
"""Transcribe an audio file to text.
Args:
audio: The audio file to transcribe.
language: Optional language hint (not currently used; auto-detected).
model: Whisper model size (legacy; ignored in dual-mode architecture).
mode: 'fast' (default) uses MLX Turbo for speed; 'accurate' uses
WhisperX with forced alignment for word-level timing.
Returns:
{
"text": "full transcription",
"segments": [ {"start": 0.0, "end": 1.5, "text": "..."}, ... ],
"language": "en",
"duration_s": 4.2,
"transcription_time_s": 0.8,
"engine": "mlx-whisper"
}
"""
import asyncio
# Save upload to a temp file (all backends need a file path)
ext = os.path.splitext(audio.filename or "audio.wav")[1] or ".wav"
tmp = tempfile.NamedTemporaryFile(delete=False, suffix=ext)
try:
content = await audio.read()
tmp.write(content)
tmp.close()
use_accurate = (mode or "").strip().lower() == "accurate"
def _run():
if use_accurate:
# Accurate mode: full WhisperX with forced alignment —
# for when the user explicitly wants word-level timing.
from services.asr_backend import get_active_asr_backend
backend = get_active_asr_backend()
result = backend.transcribe(tmp.name, word_timestamps=True)
else:
# Fast mode (default): use the fastest available engine
# (MLX Turbo on Apple Silicon). Skip word_timestamps for
# ~30% latency reduction — dictation doesn't need them.
from services.asr_backend import get_capture_asr_backend
backend = get_capture_asr_backend()
result = backend.transcribe(tmp.name, word_timestamps=False)
return result, backend.id
from services.model_manager import _gpu_pool
loop = asyncio.get_running_loop()
t0 = time.perf_counter()
result, engine_id = await loop.run_in_executor(_gpu_pool, _run)
elapsed = round(time.perf_counter() - t0, 2)
# Normalize result shape
segments = result.get("segments", [])
full_text = result.get("text", "")
if not full_text and segments:
full_text = " ".join(s.get("text", "") for s in segments).strip()
# Calculate audio duration from segments if available
duration = 0.0
if segments:
duration = max(s.get("end", 0) for s in segments)
detected_lang = result.get("language", language or "unknown")
logger.info(
"Capture transcription done: engine=%s, elapsed=%.2fs, duration=%.1fs, mode=%s",
engine_id, elapsed, duration, "accurate" if use_accurate else "fast",
)
return {
"text": full_text,
"segments": [
{
"start": round(s.get("start", 0), 2),
"end": round(s.get("end", 0), 2),
"text": s.get("text", "").strip(),
}
for s in segments
],
"language": detected_lang,
"duration_s": round(duration, 2),
"transcription_time_s": elapsed,
"engine": engine_id,
}
finally:
try:
os.unlink(tmp.name)
except OSError:
pass