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# GPU / AI-Separator Upgrade — Planning Notes (for me, the assistant)
Status: **planned, not built.** Last updated: 2026-08-23.
Context: VoiceGrab core (`voicegrab.py`) currently isolates voice with the
`noisereduce` STFT pipeline (DSP). This note covers adding a deep-learning
separator (Demucs / UVR / RoFormer family) as an optional "heavy isolation"
mode for clips with music or competing speech.
---
## 1. Problem the DSP path doesn't solve
`noisereduce` (spectral gating) is great for **stationary / broadband noise**:
fan, hum, room tone, light traffic. It **cannot** separate:
- background music (guitars, drums, other instruments overlap speech band)
- a second speaker talking
- intermittent noise bursts that contain voice-like formant structure
Spectral gate will either leave the music or carve holes in the target voice
(pumping artifacts). For AI-voice-training samples, a *source separator*
(speech vs. non-speech) is the right tool.
## 2. Candidate models (ranked for our use case)
| Model | Size (ckpt) | What it does | Notes |
|---|---|---|---|
| **Demucs `htdemucs_ft`** | ~92 MB | 4-stem (vocals/drums/bass/other) | MIT code + weights, maintained, CLI-ready, best "default choice". `htdemucs` (80 MB) is the original. Fine-tuned `_ft` preserves voice quality better — prefer it. |
| **BS-RoFormer / Mel-Band RoFormer** (vocal separation & speech enhancement) | ~50100 MB | SOTA speech/music separation quality | Checkpoint licensing is **research-leaning** — must verify before bundling/redistributing. |
| **UVR5 MDX-Net / VR Arch** checkpoints | ~70100 MB each | Community vocal-sep models, many flavors | The *app* is a GUI wrapper; the checkpoints are community-made with mixed licenses — risky to ship. Good for *reference* only. |
| **Demucs `htdemucs_6s`** | ~100 MB | 6-stem incl. a dedicated "speech" stem | Heavier; only worth it if 4-stem vocals bleed. |
Decision (revisit at build time): **`htdemucs_ft` as the default AI mode** —
permissive license, single dependency (`demucs` pulls `torch`), and its
"vocals" stem is exactly what we want. Keep an eye on RoFormer-family
license clarity; if permissive weights appear, they become the recommended
quality option.
**License gate: never bundle a checkpoint we can't point to a permissive
license for.** Demucs/`htdemucs(_ft)` = MIT. Everything else: verify first.
## 3. Hardware & performance reality check
Local machine (`voicegrab2` host): **no NVIDIA GPU** (no `nvidia-smi`),
8 cores, 14 GB RAM. So:
- Development/testing here will be **CPU-only**. That's fine for *correctness*
and pipeline integration; all real speed numbers below for "what the user
experiences" assume an RTX 30/40/50-class GPU.
- CPU fallback must still work (Windows users without GPU, Macs).
Estimates (5 s clip → 60 s clip):
| Backend | 5 s | 60 s | VRAM | Notes |
|---|---|---|---|---|
| DSP (current) | <1 s | ~1020 s | ~200 MB RAM | Always works, CPU |
| Demucs CPU | ~1025 s | ~26 min | ~12 GB RAM | usable for short samples |
| Demucs CUDA (46 GB VRAM) | ~13 s | ~1540 s | ~2 GB peak | comfortable at 6 GB+ |
| 6-stem / RoFormer | similar +3050% | similar +3050% | ~24 GB | longer context = more VRAM |
Practical guidance to surface in UI copy:
- Target clip length for AI voice samples is 30120 s → **fine on 4 GB VRAM**.
- First run downloads model (~92 MB) to user data dir. Show this explicitly.
- CPU mode: set expectations ("a few minutes for a 60 s clip").
## 4. Architecture plan
### 4.1 Backend abstraction (replace direct `VoiceReducer` calls)
```python
# voicegrab.py (sketch)
class Separator:
name: str # "None" | "Noise reduction (fast)" | "AI separation (Demucs)"
def available(self) -> tuple[bool, str]: ... # (ok, reason)
def separate(self, y: np.ndarray, sr: int, progress) -> np.ndarray: ...
class DSSeparator(Separator): # current noisereduce path, unchanged behavior
...
class DemucsSeparator(Separator): # lazy import; GPU-aware
def __init__(self):
import torch, demucs.apply, demucs.pretrained
...
def separate(self, y, sr, progress):
wav = y[None, :, :] # demucs expects (1, C, T) float32
model = self._model() # cached singleton
sources = demucs.apply.apply_model(model, wav, device=self.device, shifts=1)
vocals = sources[model.sources.index("vocals")]
return vocals[0, 0].cpu().numpy() # mono speech
```
Pipeline order becomes: **extract range → optional separation (DSP or AI) →
loudnorm → MP3 320k**. The AI path *replaces* DSP, not stacks on it — running
`noisereduce` after Demucs is usually counterproductive (double-processing
artifacts). Keep DSP as an independent mode, not a post-filter.
### 4.2 Device selection
```python
def pick_device():
if torch.cuda.is_available():
return "cuda"
if getattr(torch.backends, "mps", None) and torch.backends.mps.is_available():
return "mps" # Mac bonus, free
return "cpu"
```
Surface the active device in status text ("Using GPU (CUDA)" / "CPU mode —
will be slower").
### 4.3 GUI changes
- Replace the single "Isolate voice" checkbox with a **mode selector**:
1. `No processing` (just loudness-normalize — current non-isolate path)
2. `Noise reduction (fast)` — current DSP, default, no downloads
3. `AI voice separation (Demucs)` — shows a note: "First use downloads
~92 MB of models. GPU strongly recommended."
- Progress: `demucs.apply.apply_model` supports a callback; route it into the
existing `Worker.log` signal. On first use, `pretrained.get_model("htdemucs_ft")`
download should report progress too (it goes through `torch.hub`/`urllib` —
wrap or at least show "Downloading model… (x MB)" in the log line).
- Model cache: `appdata_dir()/models/` (already the pattern for the old
profile idea; reuse `appdata_dir()`).
### 4.4 Threading / UX
- Keep the existing `Worker(QThread)`; add a **busy spinner + cancel button**.
Demucs on CPU can run minutes — `subprocess`-style cancellation isn't
available for in-process torch, so implement a soft cancel: worker checks a
flag between `apply_model` shifts (pass `shifts=2`, check between) — or accept
no-cancel for v1 and say so.
- Disable the Export button while running (already the pattern).
## 5. Packaging — the hard part (be honest about it)
`torch` + `demucs` is the weight problem. Options, ranked:
**A. "AI Pack" as a separate installer component (recommended).**
Core app stays small & instant-start (PyInstaller onefile, ~4080 MB with
PySide6+ffmpeg). Ship a second Inno Setup *component* ("AI voice separation
(Demucs), ~1.52.5 GB after install") that drops pre-wheeled `torch`+`demucs`
wheels into `%APPDATA%\VoiceGrab\ai\wheels`. On first AI-mode use, the app
builds a venv from those wheels (`python -m venv ~/.local/share/VoiceGrab/ai_venv`)
and imports from it (or runs separation as a **sidecar process** — see B).
Pros: core users pay nothing; opt-in size; works offline after download.
Cons: most moving parts; needs careful Windows testing.
**B. Separation as a sidecar executable (cleanest runtime boundary).**
Build `separate.py``demucs-cli.exe` with PyInstaller (`--collect-all torch`
→ big, but it's an *isolated* binary, e.g. `VoiceGrab-AI.exe` next to the
core app). Core app shells out: `VoiceGrab-AI.exe --in clip.wav --out vocals.wav
--device cuda`. Pros: crash/firewall/GPU driver issues can't take down the GUI;
no venv bootstrapping; trivial to update independently; the `demucs` CLI
already does exactly this job. Cons: one more build artifact.
**I'd pick B for Windows, A-or-B for Linux.** In practice B generalizes:
it *is* the demucs CLI wrapped in a branded exe.
**C. Single onefile with torch bundled.** ❌ Don't. 8 GB+ onefile, 60120 s
extraction on every start, memory-mapped junk in `%TEMP%`. Only acceptable as
a "portable full build" niche option.
### Linux specifics
- AppImage with torch = same bloat problem → ship `VoiceGrab` (core, as today)
+ `VoiceGrab-AI` (sidecar binary, ~2 GB) or a `pip install -e .[ai]` path.
- A second `.deb` component with the wheels in `/usr/share/voicegrab-ai` is
also reasonable.
## 6. Correctness / QA plan
1. **Unit:** `separate()` returns float32 mono, shape `(T,)`, dtype invariance,
no NaN, SR unchanged (Demucs resamples to 44.1 k internally — **re-resample
back to 48 k before loudnorm**, or just feed 44.1 through to MP3; decide
and document. Voice training tools accept 44.1k fine → **simplest: feed
Demucs output at 44.1k straight to MP3**, skip the 48k round-trip).
2. **Regression:** existing `kage.mp4` 16 s test must produce a valid 320 kbps
MP3 in both DSP and AI modes; assert `ffprobe` duration ≈ 5.0 s ± 0.15 s
(MP3 encoder padding).
3. **A/B listening test:** pick 3 clips (clean speech; speech+fan;
speech+music) — currently only have `kage.mp4`; need 2 more fixtures.
Record: artifacts? "chipmunk" consonants (classic Demucs failure)?
residual music?
4. **GPU smoke:** on a CUDA machine: `torch.cuda.is_available()`, model load,
60 s clip timing, VRAM peak via `torch.cuda.max_memory_allocated()`.
Acceptance: < 2× realtime on 6 GB VRAM.
5. **CPU timeout sanity:** 60 s clip on 8-core CPU must finish < 8 min;
surface estimated time in UI ("≈ 24 min on CPU").
6. **No-GPU Windows box** (test VM): CPU path + model download + offline
second run (no re-download).
## 7. Risks / open questions
- **Model redistribution license** — only ship MIT (Demucs). Re-check
checkpoint license at build time; put a `THIRD_PARTY.md` in the installer.
- **Torch × Python 3.14** — current host is 3.14; verify wheel availability
at target versions (torch tracks newest Pythons well, but pin in CI).
- **CUDA wheel size on Windows**`torch` with cu126 wheels ≈ 2.5 GB
(nvidia-cublas etc. are separate wheels). Sidecar build must
`--collect-all torch nvidia.*`.
- **AMD/Intel users** — ROCm wheels exist (Windows + Linux) but are finicky;
IPEX for Intel Arc. v1: NVIDIA + CPU only; document the rest as "may work".
- **Demucs drift**`demucs` is PyTorch-Lightning-free now but its API
(`apply_model`, `pretrained.get_model`) has shifted across versions.
**Pin `demucs==4.1.x` in the AI-pack wheels.**
- **Voice fidelity tradeoff:** aggressive separation can thin out sibilance /
breath — exactly what TTS trainers want to *keep*. `htdemucs_ft` is the
gentler choice; also consider post-gain before loudnorm (separated vocals
sit ~610 dB lower than the mix).
- **Cancel semantics** for CPU-mode runs (see 4.4) — decide before build.
## 8. Suggested build order (if I do this)
1. `SeparationBackend` interface + move existing DSP behind it (small, no deps) — ✅ safe, half a day.
2. Add `demucs` to a **dev-only** venv extra; wire `DemucsSeparator` with
lazy import + `pick_device()`; test end-to-end on this CPU box with
`kage.mp4` (5 s, expect ~1030 s).
3. UI: mode selector + progress + device line. (Half a day.)
4. Packaging: prototype **B** (sidecar) with PyInstaller on this box (CPU
build) to shake out `--collect-all` issues; finalize Windows build script.
5. QA pass from §6; write `THIRD_PARTY.md`; bump version to 1.1.0.
Rough total: **24 focused days**, packaging being the long pole.
## 9. Files this would touch
- `voicegrab.py` — backend interface, `DemucsSeparator`, UI mode selector
- `installer/build_win.ps1` — AI sidecar build step
- `installer/VoiceGrab.iss` — new `[Components]`/`[Files]` entries
- `installer/build_linux.sh` — sidecar variant
- `requirements-ai.txt` (new) — `demucs==4.1.*`, `torch==2.*`
- `README.md` — AI mode docs + GPU guidance
- `tests/test_separators.py` (new) — §6 automation
---
*Reminder: the "big model" option is a **product feature behind a checkbox**,
never a hard dependency. Core app must stay <100 MB and start in <2 s.*

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# VoiceGrab
Cut a time range out of any video file and export a clean, loudness-normalized
**MP3 voice sample** — ready to use for AI voice training / cloning.
```
Voice / audio file: [ kage.mp4 ] [Open…]
Start: 0:10.00 Stop: 0:35.00 [Load preview] [▶ Play range]
[x] Isolate voice (reduce background noise) strength: 80%
Output MP3: [ .../kage_voicegrab.mp3 ] [Browse…]
──────────────────────────────────────────────────
[ Export MP3 ]
```
## Features
- Works with any container ffmpeg understands: mp4, mkv, mov, webm, avi, flv, m4a…
- **Isolate voice** checkbox: STFT-based noise reduction (runs on CPU, no
downloads, no GPU). Strength slider + a "stationary noise" mode for
constant fans/hum.
- Export is **320 kbps MP3**, loudness-normalized to 16 LUFS (broadcast
reference — good levels for voice-training datasets).
- "▶ Play range" lets you audition the exact segment before saving.
- Also a headless CLI (see below) for scripting/batch work.
## Requirements
- Python 3.10+
- ffmpeg on your PATH (bundled automatically in installer builds)
## Run from source
```bash
cd voicegrab2
python3 -m venv .venv
.venv/bin/pip install -r requirements.txt
.venv/bin/python voicegrab.py
```
## CLI mode
```bash
voicegrab.py --cli input.mp4 10 35 output.mp3
```
## Windows installer (single .exe + installer)
1. `pip install pyinstaller`
2. `python voicegrab.py.spec` is not needed — run:
```bat
pyinstaller --onefile --windowed --name VoiceGrab ^
--add-binary "C:\ffmpeg\bin\ffmpeg.exe;." ^
--add-binary "C:\ffmpeg\bin\ffprobe.exe;." ^
--icon assets/icon.ico voicegrab.py
```
3. Build the installer with Inno Setup:
```bat
"C:\Program Files (x86)\Inno Setup 6\ISCC.exe" installer\VoiceGrab.iss
```
(Copy `dist\VoiceGrab.exe` to `installer\` first — the script assumes
`dist\VoiceGrab.exe` relative to the project root.)
## Linux
- **AppImage**: `installer/build_linux.sh` (needs PyInstaller +
`linuxdeploy` or `appimagetool` + your ffmpeg in PATH).
- **Debian/Ubuntu**: just run the PyInstaller binary, or install ffmpeg via
apt and run from source. A `.desktop` entry template is in `installer/`.
- **Flatpak** is also a fine route if you want it in your store.
## Notes on voice-isolation quality
The built-in reducer is classic DSP (spectral gating / STFT noise
estimation) — great for fan, hum, light room tone. For music or heavy
background speech, a deep-learning model (e.g. Demucs / UVR) gives
better separation but costs ~12 GB of downloads and runs far slower
without a GPU. For voice *training* samples, the DSP route is usually
more than enough, and it keeps the installer small and startup instant.

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; VoiceGrab — Inno Setup script
; Install Inno Setup 6: https://jrsoftware.org/isdl.php
; Compile: ISCC.exe installer\VoiceGrab.iss (from project root, after build_win.ps1)
#define MyAppName "VoiceGrab"
#define MyAppVersion "1.0.0"
#define MyAppExeName "VoiceGrab.exe"
[Setup]
AppId={{B7E5A9C2-3D41-4F6B-9E0A-1C8F5D2B7A11}
AppName={#MyAppName}
AppVersion={#MyAppVersion}
AppPublisher=VoiceGrab
DefaultDirName={autopf}\VoiceGrab
DefaultGroupName=VoiceGrab
DisableProgramGroupPage=yes
OutputDir=dist
OutputBaseFilename=VoiceGrab-Setup-{#MyAppVersion}
Compression=lzma2
SolidCompression=yes
WizardStyle=modern
ArchitecturesInstallIn64BitMode=x64
ArchitecturesAllowed=x64
PrivilegesRequired=admin
PrivilegesRequiredOverridesAllowed=dialog
[Languages]
Name: "english"; MessagesFile: "compiler:Default.isl"
[Files]
Source: "dist\VoiceGrab.exe"; DestDir: "{app}"; Flags: ignoreversion
[Icons]
Name: "{group}\VoiceGrab"; Filename: "{app}\{#MyAppExeName}"
Name: "{autodesktop}\VoiceGrab"; Filename: "{app}\{#MyAppExeName}"; Tasks: desktopicon
[Tasks]
Name: "desktopicon"; Description: "{cm:CreateDesktopIcon}"; GroupDescription: "{cm:AdditionalIcons}"
[Run]
Filename: "{app}\{#MyAppExeName}"; Description: "{cm:LaunchProgram,{#StringChange(MyAppName, '&', '&&')}}"; Flags: nowait postinstall skipifsilent

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#!/usr/bin/env bash
# VoiceGrab — Linux build (produces dist/VoiceGrab-<arch> binary; AppImage if linuxdeploy available)
set -euo pipefail
cd "$(dirname "$0")/.."
PY="${PYTHON:-.venv/bin/python}"
[ -x "$PY" ] || { echo "Create the venv first: python3 -m venv .venv && .venv/bin/pip install -r requirements.txt"; exit 1; }
command -v ffmpeg >/dev/null || { echo "ffmpeg not found in PATH (needed to resolve the binary)"; exit 1; }
"$PY" -m pip install pyinstaller
FFMPEG_BIN="$(command -v ffmpeg)"
FFPROBE_BIN="$(command -v ffprobe || true)"
EXTRA=()
if [ -n "${FFPROBE_BIN:-}" ]; then
EXTRA+=(--add-binary "$FFPROBE_BIN:.")
fi
"$PY" -m PyInstaller --noconfirm --onefile --windowed --name VoiceGrab \
--add-binary "$FFMPEG_BIN:." \
"${EXTRA[@]}" \
--add-data "assets/icon.png:assets" \
--icon assets/icon.ico \
voicegrab.py
echo
echo "Binary: dist/VoiceGrab"
echo "Test: ./dist/VoiceGrab"
# Optional: wrap in an AppImage
if command -v linuxdeploy >/dev/null 2>&1; then
echo "Building AppImage with linuxdeploy..."
linuxdeploy --appimage --input dist/ --output appimage
elif command -v appimagetool >/dev/null 2>&1; then
echo "linuxdeploy not found; appimagetool present — build a .dir first, then run:"
echo " appimagetool dist/VoiceGrab-Dir VoiceGrab-x86_64.AppImage"
else
echo "Tip: install linuxdeploy (or appimagetool) to produce an AppImage."
fi

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# VoiceGrab — Windows build script
# Usage (from project root): powershell -ExecutionPolicy Bypass -File installer\build_win.ps1
$ErrorActionPreference = "Stop"
$root = Split-Path -Parent $PSScriptRoot
Set-Location $root
# 1. Ensure a venv with deps
if (-not (Test-Path ".venv\Scripts\python.exe")) {
python -m venv .venv
}
& ".venv\Scripts\python.exe" -m pip install --upgrade pip
& ".venv\Scripts\python.exe" -m pip install -r requirements.txt pyinstaller
# 2. Grab a static ffmpeg build (BtbN build — includes libmp3lame)
$ffdir = "third_party\ffmpeg\bin"
if (-not (Test-Path "$ffdir\ffmpeg.exe")) {
New-Item -ItemType Directory -Force -Path $ffdir | Out-Null
$zip = "third_party\ffmpeg.zip"
$url = "https://github.com/BtbN/FFmpeg-Builds/releases/latest/download/ffmpeg-master-latest-win64-gpl.zip"
Write-Host "Downloading ffmpeg (one-time)..."
Invoke-WebRequest -Uri $url -OutFile $zip
Expand-Archive $zip -DestinationPath "third_party\ffmpeg_tmp"
$found = Get-ChildItem -Recurse "third_party\ffmpeg_tmp" -Filter "ffmpeg.exe" | Select-Object -First 1
Copy-Item $found.FullName "$ffdir\"
$probe = Join-Path $found.DirectoryName "ffprobe.exe"
if (Test-Path $probe) { Copy-Item $probe "$ffdir\" }
Remove-Item $zip -Force; Remove-Item "third_party\ffmpeg_tmp" -Recurse -Force
}
# 3. PyInstaller
& ".venv\Scripts\pyinstaller.exe" --noconfirm --onefile --windowed --name VoiceGrab `
--add-binary "$ffdir\ffmpeg.exe;." `
--add-binary "$ffdir\ffprobe.exe;." `
--add-data "assets\icon.png;assets" `
--icon "assets\icon.ico" `
voicegrab.py
Write-Host "Built dist\VoiceGrab.exe"
Write-Host ""
Write-Host "Next: build the installer with Inno Setup:"
Write-Host ' "C:\Program Files (x86)\Inno Setup 6\ISCC.exe" installer\VoiceGrab.iss'

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[Desktop Entry]
Type=Application
Name=VoiceGrab
Comment=Cut a time range from a video and export a clean MP3 voice sample
Exec=voicegrab
Icon=voicegrab
Categories=AudioVideo;Audio;
Keywords=audio;mp3;voice;clip;

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PySide6>=6.6
numpy
scipy
soundfile
noisereduce

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#!/usr/bin/env python3
"""VoiceGrab — cut a time range from a video file and export a clean MP3 voice sample.
- Pick an input video (mp4, mkv, webm, avi, mov, ...)
- Set start / stop timestamps
- Optional: isolate voice with noise reduction (no GPU or big models needed)
- Export MP3 (320 kbps, loudness-normalized to -16 LUFS good for AI voice training)
- Waveform preview with the selected range highlighted
"""
from __future__ import annotations
import os
import platform
import shutil
import subprocess
import sys
import tempfile
import wave
APP_NAME = "VoiceGrab"
APP_VERSION = "1.0.0"
# ---------------------------------------------------------------------------
# Resource / executable helpers
# ---------------------------------------------------------------------------
def app_dir() -> str:
"""Directory that contains our files (bundled data dir in frozen builds)."""
if getattr(sys, "frozen", False):
return sys._MEIPASS # type: ignore[attr-defined]
return os.path.dirname(os.path.abspath(__file__))
def appdata_dir() -> str:
"""Per-user directory for saved model noise profiles."""
if platform.system() == "Windows":
base = os.environ.get("APPDATA", os.path.expanduser("~"))
elif platform.system() == "Darwin":
base = os.path.expanduser("~/Library/Application Support")
else:
base = os.environ.get("XDG_DATA_HOME", os.path.expanduser("~/.local/share"))
d = os.path.join(base, APP_NAME)
os.makedirs(d, exist_ok=True)
return d
# ---------------------------------------------------------------------------
# ffmpeg helpers (bundled binary in frozen builds, system ffmpeg otherwise)
# ---------------------------------------------------------------------------
def ffmpeg_exe() -> str:
if getattr(sys, "frozen", False):
exe = os.path.join(app_dir(), "ffmpeg.exe" if platform.system() == "Windows" else "ffmpeg")
if os.path.exists(exe):
# ensure executable bit
try:
os.chmod(exe, 0o755)
except OSError:
pass
return exe
return "ffmpeg"
def ffprobe_exe() -> str:
if getattr(sys, "frozen", False):
exe = os.path.join(app_dir(), "ffprobe.exe" if platform.system() == "Windows" else "ffprobe")
if os.path.exists(exe):
try:
os.chmod(exe, 0o755)
except OSError:
pass
return exe
return "ffprobe"
class FfmpegError(RuntimeError):
pass
def probe_duration(path: str) -> float:
try:
out = subprocess.run(
[ffprobe_exe(), "-hide_banner", "-v", "error",
"-show_entries", "format=duration", "-of", "default=nw=1:nk=1", path],
capture_output=True, text=True, timeout=60,
)
return float(out.stdout.strip().splitlines()[0])
except (ValueError, IndexError, subprocess.SubprocessError) as exc:
raise FfmpegError(f"Could not read media file: {path}") from exc
def extract_wav(path: str, start: float, stop: float, workdir: str) -> str:
"""Extract the selected range as mono 48 kHz WAV (PCM 16-bit)."""
out = os.path.join(workdir, "clip.wav")
cmd = [
ffmpeg_exe(), "-hide_banner", "-v", "error", "-y",
"-ss", f"{start:.3f}", "-to", f"{stop:.3f}",
"-i", path,
"-vn", "-ac", "1", "-ar", "48000",
"-c:a", "pcm_s16le", out,
]
r = subprocess.run(cmd, capture_output=True, text=True, timeout=600)
if r.returncode != 0 or not os.path.exists(out):
raise FfmpegError(f"Audio extraction failed:\n{r.stderr.strip()[:1500]}")
return out
def wav_stats(path: str):
"""Return (nframes, samplerate, rms) of a 16-bit WAV."""
with wave.open(path, "rb") as w:
n = w.getnframes()
sr = w.getframerate()
data = w.readframes(n)
import array
a = array.array("h")
a.frombytes(data[: len(data) // 2 * 2])
if len(a) == 0:
return n, sr, 0.0
rms = (sum((x / 32768.0) ** 2 for x in a) / len(a)) ** 0.5
return n, sr, float(rms)
def peaks(path: str, buckets: int = 2000):
"""Downsample a 16-bit WAV to (max, min) per bucket for plotting."""
with wave.open(path, "rb") as w:
n = w.getnframes()
data = w.readframes(n)
import array
a = array.array("h")
a.frombytes(data[: len(data) // 2 * 2])
m = len(a)
if m == 0:
return [], []
step = max(1, m // buckets)
maxs, mins = [], []
for i in range(0, m, step):
chunk = a[i:i + step]
maxs.append(max(chunk) / 32768.0)
mins.append(min(chunk) / 32768.0)
return maxs, mins
# ---------------------------------------------------------------------------
# Voice isolation (lightweight, CPU-only)
# ---------------------------------------------------------------------------
try:
import numpy as np
import noisereduce as nr
HAVE_NR = True
NR_IMPORT_ERROR = ""
except Exception as _exc: # pragma: no cover
HAVE_NR = False
NR_IMPORT_ERROR = str(_exc)
class VoiceReducer:
"""STFT noise reduction with an optional noise reference clip."""
def __init__(self, intensity: float, stationary: bool):
self.intensity = float(intensity) # 0.0 .. 1.0
self.stationary = stationary
self.noise_ref = None # optional numpy array of noise
def set_noise_ref(self, noise):
self.noise_ref = noise
def reduce(self, y: "np.ndarray", sr: int) -> "np.ndarray":
kwargs = dict(
y=y, sr=sr,
prop_decrease=min(0.95, self.intensity),
stationary=self.stationary,
use_tqdm=False,
)
if self.noise_ref is not None:
kwargs["y_noise"] = self.noise_ref
return nr.reduce_noise(**kwargs)
# ---------------------------------------------------------------------------
# Audio IO helpers (numpy/soundfile)
# ---------------------------------------------------------------------------
def read_wav_f32(path: str):
import soundfile as sf
data, sr = sf.read(path, dtype="float32")
return data, sr
def write_wav_f32(path: str, y, sr: int):
import soundfile as sf
sf.write(path, y, sr, subtype="PCM_16")
# ---------------------------------------------------------------------------
# GUI
# ---------------------------------------------------------------------------
def _import_pyside():
from PySide6 import QtCore, QtGui, QtWidgets # noqa: F401
return QtCore, QtGui, QtWidgets
def run_gui() -> int:
QtCore, QtGui, QtWidgets = _import_pyside()
def _normalize(y):
import numpy as _np
p = _np.percentile(_np.abs(y), 99.5)
if p < 1e-6:
return y
return (y / p * 0.891) # ~ -1 dBFS reference peak
class Worker(QtCore.QThread):
log = QtCore.Signal(str)
done = QtCore.Signal(object, str) # (success, message)
def __init__(self, job):
super().__init__()
self.job = job
def run(self):
import numpy as _np
try:
self.log.emit("Extracting audio range…")
with tempfile.TemporaryDirectory(prefix="voicegrab-") as wd:
wav = extract_wav(self.job["input"], self.job["start"], self.job["stop"], wd)
y, sr = read_wav_f32(wav)
if self.job["isolate"]:
if not HAVE_NR:
raise FfmpegError(f"Voice isolation unavailable: {NR_IMPORT_ERROR}")
self.log.emit("Isolating voice (noise reduction)\u2026")
red = VoiceReducer(self.job["intensity"], self.job["stationary"])
y = red.reduce(y, sr)
else:
self.log.emit("Normalizing loudness…")
y = _normalize(y)
work_wav = os.path.join(wd, "work.wav")
write_wav_f32(work_wav, y, sr)
self.log.emit("Encoding MP3 (320 kbps, loudness-normalized)…")
enc = [
ffmpeg_exe(), "-hide_banner", "-v", "error", "-y",
"-i", work_wav,
"-af", f"loudnorm=I={self.job['lufts']}:TP=-1.5:LRA=11",
"-c:a", "libmp3lame", "-b:a", "320k", self.job["output"],
]
r = subprocess.run(enc, capture_output=True, text=True, timeout=300)
if r.returncode != 0 or not os.path.exists(self.job["output"]):
raise FfmpegError(f"MP3 encoding failed:\n{r.stderr.strip()[:1500]}")
self.log.emit("Done.")
self.done.emit(True, self.job["output"])
except Exception as exc:
self.done.emit(False, str(exc))
return _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize)
def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
app = QtWidgets.QApplication(sys.argv)
app.setApplicationName(APP_NAME)
app.setApplicationVersion(APP_VERSION)
class MainWindow(QtWidgets.QMainWindow):
def __init__(self):
super().__init__()
self.setWindowTitle(f"{APP_NAME} — MP3 voice sample cutter")
self.resize(920, 640)
self.input_path = None
self.duration = 0.0
self.worker = None
self._build()
# ---------- UI ----------
def _build(self):
central = QtWidgets.QWidget()
self.setCentralWidget(central)
lay = QtWidgets.QVBoxLayout(central)
lay.setContentsMargins(14, 14, 14, 14)
lay.setSpacing(12)
# Row 1: input file
row1 = QtWidgets.QHBoxLayout()
row1.addWidget(QtWidgets.QLabel("Video / audio file:"))
self.in_edit = QtWidgets.QLineEdit()
self.in_edit.setPlaceholderText("Choose an .mp4 / .mkv / .mov / .webm … file")
row1.addWidget(self.in_edit, 1)
btn_open = QtWidgets.QPushButton("Open…")
btn_open.clicked.connect(self.choose_input)
row1.addWidget(btn_open)
self.lbl_info = QtWidgets.QLabel("")
self.lbl_info.setStyleSheet("color:#64748b;")
row1.addWidget(self.lbl_info)
lay.addLayout(row1)
# Row 2: times
row2 = QtWidgets.QHBoxLayout()
row2.addWidget(QtWidgets.QLabel("Start:"))
self.t_start = QtWidgets.QLineEdit("0:00.00")
self.t_start.setFixedWidth(110)
row2.addWidget(self.t_start)
row2.addSpacing(12)
row2.addWidget(QtWidgets.QLabel("Stop:"))
self.t_stop = QtWidgets.QLineEdit("0:00.00")
self.t_stop.setFixedWidth(110)
row2.addWidget(self.t_stop)
self.lbl_range = QtWidgets.QLabel("")
self.lbl_range.setStyleSheet("color:#64748b;")
row2.addWidget(self.lbl_range)
row2.addStretch(1)
btn_preview = QtWidgets.QPushButton("Load preview")
btn_preview.setToolTip("Extract the selected range so you can listen before saving")
btn_preview.clicked.connect(self.load_preview)
row2.addWidget(btn_preview)
btn_listen = QtWidgets.QPushButton("▶ Play range")
btn_listen.clicked.connect(self.play_range)
row2.addWidget(btn_listen)
lay.addLayout(row2)
# Row 3: isolation + output
self.chk_isolate = QtWidgets.QCheckBox("Isolate voice (reduce background noise)")
self.chk_isolate.setChecked(True)
self.chk_isolate.toggled.connect(self._isolate_toggled)
lay.addWidget(self.chk_isolate)
iso_row = QtWidgets.QHBoxLayout()
iso_row.addWidget(QtWidgets.QLabel("Noise reduction strength:"))
self.spn_intensity = QtWidgets.QSpinBox()
self.spn_intensity.setRange(0, 100)
self.spn_intensity.setValue(80)
self.spn_intensity.setSuffix("%")
iso_row.addWidget(self.spn_intensity)
iso_row.addSpacing(16)
self.chk_stationary = QtWidgets.QCheckBox(
"Stationary noise (fan / hum — better if constant)")
iso_row.addWidget(self.chk_stationary)
iso_row.addStretch(1)
self.lbl_iso_status = QtWidgets.QLabel("")
self.lbl_iso_status.setStyleSheet("color:#64748b;")
iso_row.addWidget(self.lbl_iso_status)
lay.addLayout(iso_row)
self.iso_row_widget = iso_row
out_row = QtWidgets.QHBoxLayout()
out_row.addWidget(QtWidgets.QLabel("Output MP3:"))
self.out_edit = QtWidgets.QLineEdit()
self.out_edit.setPlaceholderText("Defaults to <input>_voicegrab.mp3 next to the source")
out_row.addWidget(self.out_edit, 1)
btn_out = QtWidgets.QPushButton("Browse…")
btn_out.clicked.connect(self.choose_output)
out_row.addWidget(btn_out)
lay.addLayout(out_row)
# Waveform
lay.addWidget(QtWidgets.QLabel("Waveform of the selected range:"))
self.wave = WaveView()
self.wave.setMinimumHeight(160)
lay.addWidget(self.wave, 1)
# Bottom bar
bar = QtWidgets.QHBoxLayout()
self.btn_export = QtWidgets.QPushButton("Export MP3")
self.btn_export.setMinimumHeight(40)
self.btn_export.setStyleSheet(
"font-size:14px; font-weight:600; background:#0ea5e9; color:white;"
"border-radius:6px; padding:4px 18px;")
self.btn_export.clicked.connect(self.export)
bar.addWidget(self.btn_export)
self.lbl_status = QtWidgets.QLabel("Ready.")
self.lbl_status.setStyleSheet("color:#64748b;")
bar.addWidget(self.lbl_status, 1)
lay.addLayout(bar)
if not HAVE_NR:
self.lbl_iso_status.setText(f"⚠ voice isolation unavailable ({NR_IMPORT_ERROR})")
self.chk_isolate.setEnabled(False)
# ---------- helpers ----------
def _isolate_toggled(self, on):
for w in self.iso_row_widget.items():
if isinstance(w, QtWidgets.QWidget) and w not in (self.lbl_iso_status,):
w.setEnabled(on)
@staticmethod
def _parse_time(s: str) -> float:
"""Accept 125, 1:05, 1:05.5, 1:05:00, 01:00:00.500."""
s = s.strip()
parts = s.split(":")
if len(parts) > 3:
raise ValueError("Invalid time format")
t = 0.0
for i, p in enumerate(parts):
v = float(p)
t = t * 60 + v
return max(0.0, t)
def _fmt_time(self, t: float) -> str:
m, s = divmod(t, 60)
h, m = divmod(int(m), 60)
if h:
return f"{h}:{int(m):02d}:{s:05.2f}"
return f"{int(m)}:{s:05.2f}"
def _times(self):
start = self._parse_time(self.t_start.text() or "0")
stop = self._parse_time(self.t_stop.text() or "0")
if start >= stop:
raise ValueError("Start must be before stop.")
return start, stop
def refresh_info(self):
if self.input_path and self.duration:
self.lbl_info.setText(
f"Duration {self._fmt_time(self.duration)} Range {self._range_text()}")
else:
self.lbl_info.setText("")
def _range_text(self):
try:
s, e = self._times()
return f"{s:.2f}s {e:.2f}s (clip {e - s:.2f}s)"
except ValueError:
return "— invalid range —"
# ---------- slots ----------
def choose_input(self):
path, _ = QtWidgets.QFileDialog.getOpenFileName(
self, "Choose video or audio file", "",
"Media files (*.mp4 *.mkv *.mov *.webm *.avi *.m4a *.mp3 *.wav *.ogg *.flv *.ts);;All files (*)")
if not path:
return
self.input_path = path
self.in_edit.setText(path)
self.t_stop.setText(self._fmt_time(self.duration)) # after probe below
try:
self.duration = probe_duration(path)
self.t_stop.setText(self._fmt_time(self.duration))
except FfmpegError as exc:
self.status(str(exc)); return
self.refresh_info()
self.load_preview()
def choose_output(self):
if not self.input_path:
return
base = os.path.splitext(os.path.basename(self.input_path))[0]
start = os.path.join(os.path.dirname(self.input_path), base + "_voicegrab.mp3")
path, _ = QtWidgets.QFileDialog.getSaveFileName(
self, "Save MP3 as", start, "MP3 files (*.mp3)")
if path:
self.out_edit.setText(path)
def load_preview(self):
if not self.input_path:
return
try:
s, e = self._times()
except ValueError as exc:
self.status(str(exc)); return
try:
with tempfile.TemporaryDirectory(prefix="voicegrab-") as wd:
wav = extract_wav(self.input_path, s, e, wd)
maxs, mins = peaks(wav)
n, sr, rms = wav_stats(wav)
except FfmpegError as exc:
self.status(str(exc)); return
self.wave.set_peaks(maxs, mins)
self.lbl_range.setText(f"clip {n / sr:.2f}s @ {sr} Hz RMS {rms:.3f}")
def play_range(self):
if not self.input_path:
return
try:
s, e = self._times()
except ValueError as exc:
self.status(str(exc)); return
player = os.path.join(tempfile.gettempdir(), "voicegrab_preview.mp3")
try:
cmd = [
ffmpeg_exe(), "-hide_banner", "-v", "error", "-y",
"-ss", f"{s:.3f}", "-to", f"{e:.3f}", "-i", self.input_path,
"-vn", "-c:a", "libmp3lame", "-b:a", "128k", player,
]
subprocess.run(cmd, check=True, capture_output=True, text=True, timeout=300)
if platform.system() == "Windows":
os.startfile(player) # type: ignore[attr-defined]
elif platform.system() == "Darwin":
subprocess.Popen(["open", player])
else:
subprocess.Popen(["xdg-open", player])
self.status("Playing range in system player…")
except (subprocess.SubprocessError, OSError) as exc:
self.status(f"Could not play range: {exc}")
def status(self, msg: str):
self.lbl_status.setText(msg)
def export(self):
if self.worker is not None:
self.status("Already working…"); return
if not self.input_path:
QtWidgets.QMessageBox.warning(self, APP_NAME, "Choose an input file first.")
return
try:
s, e = self._times()
except ValueError as exc:
QtWidgets.QMessageBox.warning(self, APP_NAME, str(exc)); return
out = self.out_edit.text().strip()
if not out:
base = os.path.splitext(os.path.basename(self.input_path))[0]
out = os.path.join(os.path.dirname(os.path.abspath(self.input_path)),
base + "_voicegrab.mp3")
self.btn_export.setEnabled(False)
self.status("Working… (see log in status area)")
self.worker = Worker({
"input": self.input_path, "start": s, "stop": e,
"output": out,
"isolate": self.chk_isolate.isChecked(),
"intensity": self.spn_intensity.value() / 100.0,
"stationary": self.chk_stationary.isChecked(),
"lufts": -16,
})
self.worker.log.connect(self.status)
self.worker.done.connect(self._export_done)
self.worker.start()
def _export_done(self, ok, msg):
self.btn_export.setEnabled(True)
self.worker = None
if ok:
self.status(f"✔ Saved: {msg}")
QtWidgets.QMessageBox.information(
self, APP_NAME,
f"MP3 saved to:\n{msg}\n\nTip: 30120 s of clean speech is usually "
f"enough for high-quality AI voice cloning.")
else:
self.status(f"{msg.splitlines()[0]}")
QtWidgets.QMessageBox.critical(self, APP_NAME, msg)
# ---------------- wave view ----------------
class WaveView(QtWidgets.QWidget):
def __init__(self):
super().__init__()
self.maxs: list = []
self.mins: list = []
self.setMinimumSize(120, 80)
def set_peaks(self, maxs, mins):
self.maxs = list(maxs)
self.mins = list(mins)
self.update()
def paintEvent(self, ev):
with QtGui.QPainter(self) as p:
w, h = self.width(), self.height()
mid = h / 2
p.fillRect(0, 0, w, h, QtGui.QColor("#0f172a"))
p.setPen(QtGui.QColor("#334155"))
p.drawLine(0, int(mid), w, int(mid))
if not self.maxs:
p.setPen(QtGui.QColor("#94a3b8"))
p.drawText(0, 0, w, h, int(QtCore.Qt.AlignCenter), "Open a file to see the waveform")
return
n = len(self.maxs)
color = QtGui.QColor("#38bdf8")
for i, (mx, mn) in enumerate(zip(self.maxs, self.mins)):
x = i * w / n
y1 = mid - mx * (mid - 4)
y2 = mid - mn * (mid - 4)
p.setPen(color)
p.drawLine(int(x), int(y1), int(x), max(int(y1), int(y2)))
win = MainWindow()
win.show()
return app.exec()
if __name__ == "__main__":
if "--cli" in sys.argv:
# Simple headless mode: voicegrab --cli <input> <start> <stop> [output.mp3]
sys.argv = [a for a in sys.argv if a != "--cli"]
if len(sys.argv) < 4:
print("usage: voicegrab --cli <input> <start> <stop> [output.mp3]")
raise SystemExit(2)
inp, s, e = sys.argv[1], float(sys.argv[2]), float(sys.argv[3])
out = sys.argv[4] if len(sys.argv) > 4 else os.path.join(
os.path.dirname(os.path.abspath(inp)),
os.path.splitext(os.path.basename(inp))[0] + "_voicegrab.mp3")
import numpy as _np
with tempfile.TemporaryDirectory(prefix="voicegrab-") as wd:
wav = extract_wav(inp, s, e, wd)
y, sr = read_wav_f32(wav)
if HAVE_NR:
red = VoiceReducer(0.8, stationary=True)
y = red.reduce(y, sr)
red_wav = os.path.join(wd, "reduced.wav")
write_wav_f32(red_wav, y, sr)
subprocess.run([
ffmpeg_exe(), "-hide_banner", "-v", "error", "-y",
"-i", red_wav, "-af", "loudnorm=I=-16:TP=-1.5:LRA=11",
"-c:a", "libmp3lame", "-b:a", "320k", out,
], check=True)
print(f"Saved {out}")
else:
raise SystemExit(run_gui())