Add AI voice separation (Demucs) as toggleable mode, v1.1.0

Implements the GPU-upgrade.md plan:

- Separator backend interface (DSSeparator = existing noisereduce path,
  DemucsSeparator = htdemucs_ft AI separation) — one mode at a time,
  never stacked
- DemucsSeparator: lazy imports (core app stays dep-free), device
  auto-pick cuda -> mps -> cpu, model cache under appdata/models,
  chunk progress reporting, soft cancel between chunks
- GUI: 'Isolate voice' checkbox replaced by a 3-way mode selector
  (fast DSP / AI Demucs / none), DSP strength controls follow the
  selection, AI radio disabled with guidance when torch+demucs are
  missing, Cancel button for in-flight AI runs
- CLI: --mode none|dsp|ai (default dsp, unchanged behavior)
- requirements-ai.txt (torch 2.x + demucs 4.1.x), THIRD_PARTY.md,
  README updates, optional AI pack flags in installer scripts,
  version bumped to 1.1.0
- tests/test_separators.py: 10 tests incl. graceful-degradation and
  full-pipeline regression (5.0 s ± 0.15 s, ~320 kbps) in all modes

.gitignore

@@ -0,0 +1,6 @@
+.venv/
+__pycache__/
+*.pyc
+dist/
+build/
+third_party/

README.md

@@ -6,25 +6,39 @@ Cut a time range out of any video file and export a clean, loudness-normalized
 ```
 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%
+Voice processing (choose one):
+  (•) Fast noise reduction (CPU) — fan / hum / room tone; no downloads
+  ( ) AI voice separation (Demucs) — music & second speakers; GPU if available
+  ( ) No processing — just loudness-normalize
+  strength: 80%   [x] stationary noise
 Output MP3: [ .../kage_voicegrab.mp3            ] [Browse…]
 ──────────────────────────────────────────────────
-[ Export MP3 ]
+[ Cancel ] [ 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
+- **Voice processing — pick one mode** (toggle, mutually exclusive):
-  downloads, no GPU). Strength slider + a "stationary noise" mode for
+  - **Fast noise reduction (CPU, default)** — STFT spectral gating via
-  constant fans/hum.
+    `noisereduce`. Great for fan, hum, room tone. No downloads, no GPU;
+    strength slider + a "stationary noise" mode for constant fans/hum.
+  - **AI voice separation (Demucs)** — deep-learning 4-stem separation
+    (we keep the *vocals* stem). Handles background music and a second
+    speaker, which spectral gating cannot. Optional install; uses an
+    NVIDIA GPU (CUDA) or Apple GPU if present, otherwise CPU (slower).
+  - **No processing** — just loudness-normalize.
 - 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.
+- AI runs report chunk progress in the status bar and can be **cancelled**
+  between chunks.
 - Also a headless CLI (see below) for scripting/batch work.
 
 ## Requirements
 - Python 3.10+
 - ffmpeg on your PATH (bundled automatically in installer builds)
+- Optional, only for AI voice separation: `pip install -r requirements-ai.txt`
+  (pulls `torch` + `demucs`, ~1.5–2.5 GB with CUDA wheels)
 
 ## Run from source
 ```bash
@@ -34,11 +48,44 @@ python3 -m venv .venv
 .venv/bin/python voicegrab.py
 ```
 
+## AI voice separation (optional — GPU recommended)
+```bash
+.venv/bin/pip install -r requirements-ai.txt
+```
+- Pick **AI voice separation (Demucs)** in the GUI. The mode is disabled
+  (with an explanation) until `torch` + `demucs` are installed — the core
+  app never requires them.
+- **First use downloads the `htdemucs_ft` model (~90 MB)** to the app data
+  dir (`<appdata>/VoiceGrab/models/`) and caches it — later runs are
+  instant. GPU (NVIDIA CUDA / Apple) is used automatically when present,
+  otherwise CPU.
+- Performance expectations (5 s → 60 s clip):
+  - GPU (≥ 6 GB VRAM): ~1–3 s → ~15–40 s
+  - CPU (8 cores): ~10–25 s → a few minutes
+- **Cancel** stops an AI run at the next chunk boundary. Fast DSP runs
+  finish in seconds and are not cancellable (by design).
+- Quality note: aggressive separation can thin sibilance/breath — for voice
+  *training* samples the fast DSP mode is usually enough; reach for AI mode
+  when there's music or a competing speaker.
+- Licensing: Demucs code is MIT and the `htdemucs_ft` weights are MIT
+  (see `THIRD_PARTY.md`).
+
 ## CLI mode
 ```bash
-voicegrab.py --cli input.mp4 10 35 output.mp3
+voicegrab.py --cli input.mp4 10 35 output.mp3              # default: fast DSP
+voicegrab.py --cli input.mp4 10 35 output.mp3 --mode dsp   # fast noise reduction
+voicegrab.py --cli input.mp4 10 35 output.mp3 --mode ai    # Demucs (requirements-ai.txt)
+voicegrab.py --cli input.mp4 10 35 output.mp3 --mode none  # just loudness-normalize
 ```
 
+## Tests
+```bash
+.venv/bin/python tests/test_separators.py   # or: .venv/bin/pytest tests/
+```
+Covers: DSP shape/dtype/NaN invariants, AI graceful degradation without
+torch, device selection, AI unit separation (44.1 kHz output), and the full
+CLI pipeline in all three modes (valid 320 kbps MP3, duration 5.0 s ± 0.15 s).
+
 ## Windows installer (single .exe + installer)
 1. `pip install pyinstaller`
 2. `python voicegrab.py.spec` is not needed — run:
@@ -55,17 +102,26 @@ voicegrab.py --cli input.mp4 10 35 output.mp3
    (Copy `dist\VoiceGrab.exe` to `installer\` first — the script assumes
    `dist\VoiceGrab.exe` relative to the project root.)
 
+   To ship the AI mode inside the frozen exe (bigger, ~2 GB), install
+   `requirements-ai.txt` into the venv first and add
+   `--collect-all demucs --collect-all torch` to the PyInstaller command.
+
 ## 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.
+- AI mode: `WITH_AI=1 ./installer/build_linux.sh` bundles `demucs`/`torch`
+  into the binary (bigger build).
 
 ## Notes on voice-isolation quality
-The built-in reducer is classic DSP (spectral gating / STFT noise
+- **Fast mode** is classic DSP (spectral gating / STFT noise estimation) —
-estimation) — great for fan, hum, light room tone. For music or heavy
+  great for fan, hum, light room tone; it can't remove music or a second
-background speech, a deep-learning model (e.g. Demucs / UVR) gives
+  speaker.
-better separation but costs ~1–2 GB of downloads and runs far slower
+- **AI mode** (Demucs `htdemucs_ft`) is a source separator — the right tool
-without a GPU. For voice *training* samples, the DSP route is usually
+  for speech-over-music or competing voices. It's a product feature behind a
-more than enough, and it keeps the installer small and startup instant.
+  radio button, never a hard dependency: the core app stays small and starts
+  in < 2 s without `torch`/`demucs` installed.
+- The AI path *replaces* the DSP path — one or the other, never stacked
+  (double-processing adds artifacts).

THIRD_PARTY.md

@@ -0,0 +1,27 @@
+# Third-Party Components
+
+## Core (always)
+
+| Component | License | Notes |
+|---|---|---|
+| PySide6 (Qt for Python) | LGPL v3 / GPL | GUI toolkit |
+| numpy, scipy, soundfile | BSD / BSD-3 / 3-clause BSD | audio math & IO |
+| noisereduce | Unlicense / MIT (see repo) | DSP voice isolation ("fast" mode) |
+| ffmpeg / ffprobe | LGPL v2.1+ (GPL builds exist) | media I/O; bundled binary must match license |
+
+## Optional — AI voice separation (only if the user installs `requirements-ai.txt`)
+
+| Component | License | Notes |
+|---|---|---|
+| demucs 4.1.x (Meta/Facebook) | Code: MIT | source separation engine |
+| PyTorch (torch 2.x) | BSD-3 | deep-learning runtime |
+| htdemucs_ft weights (~90 MB) | **MIT** (per the model card of `adefossez/HTDemucs-ft` on Hugging Face) | downloaded on first use from Hugging Face, cached in the user data dir — never bundled |
+| huggingface_hub | Apache-2.0 | model download plumbing |
+
+### License gate (project policy)
+
+We only use checkpoints whose license we can point to. Demucs code = MIT,
+`htdemucs_ft` weights = MIT (model card). They are only ever *downloaded by*
+the *user* on first use — never bundled by us. RoFormer / UVR community
+checkpoints have mixed or research-only licensing and are therefore **not**
+used.

installer/VoiceGrab.iss

@@ -3,7 +3,7 @@
 ; Compile:  ISCC.exe installer\VoiceGrab.iss   (from project root, after build_win.ps1)
 
 #define MyAppName "VoiceGrab"
-#define MyAppVersion "1.0.0"
+#define MyAppVersion "1.1.0"
 #define MyAppExeName "VoiceGrab.exe"
 
 [Setup]

installer/build_linux.sh

@@ -1,5 +1,6 @@
 #!/usr/bin/env bash
 # VoiceGrab — Linux build (produces dist/VoiceGrab-<arch> binary; AppImage if linuxdeploy available)
+#   WITH_AI=1 ./installer/build_linux.sh   → also bundle torch+demucs (bigger build)
 set -euo pipefail
 cd "$(dirname "$0")/.."
 
@@ -8,6 +9,13 @@ PY="${PYTHON:-.venv/bin/python}"
 command -v ffmpeg >/dev/null || { echo "ffmpeg not found in PATH (needed to resolve the binary)"; exit 1; }
 
 "$PY" -m pip install pyinstaller
+if [ "${WITH_AI:-0}" = "1" ]; then
+  echo "Building with AI voice separation (torch+demucs)…"
+  "$PY" -m pip install -r requirements-ai.txt
+  AI_FLAGS=(--collect-all demucs --collect-all torch)
+else
+  AI_FLAGS=()
+fi
 
 FFMPEG_BIN="$(command -v ffmpeg)"
 FFPROBE_BIN="$(command -v ffprobe || true)"
@@ -20,6 +28,7 @@ fi
 "$PY" -m PyInstaller --noconfirm --onefile --windowed --name VoiceGrab \
   --add-binary "$FFMPEG_BIN:." \
   "${EXTRA[@]}" \
+  "${AI_FLAGS[@]}" \
   --add-data "assets/icon.png:assets" \
   --icon assets/icon.ico \
   voicegrab.py

installer/build_win.ps1

@@ -1,5 +1,10 @@
 # VoiceGrab — Windows build script
-# Usage (from project root):  powershell -ExecutionPolicy Bypass -File installer\build_win.ps1
+# Usage (from project root):  powershell -ExecutionPolicy Bypass -File installer\build_win.ps1 [-BuildAI]
+#   -BuildAI   also bundle the optional AI voice separation (torch+demucs;
+#              exe grows by ~2 GB). Core users don't need it.
+param(
+    [switch]$BuildAI
+)
 $ErrorActionPreference = "Stop"
 $root = Split-Path -Parent $PSScriptRoot
 Set-Location $root
@@ -10,6 +15,16 @@ if (-not (Test-Path ".venv\Scripts\python.exe")) {
 }
 & ".venv\Scripts\python.exe" -m pip install --upgrade pip
 & ".venv\Scripts\python.exe" -m pip install -r requirements.txt pyinstaller
+if ($BuildAI) {
+    Write-Host "Building with AI voice separation (torch+demucs)…"
+    & ".venv\Scripts\python.exe" -m pip install -r requirements-ai.txt
+}
+
+# AI-mode collection flags (no-op when not building the AI pack)
+$AI_FLAGS = @()
+if ($BuildAI) {
+    $AI_FLAGS = @("--collect-all", "demucs", "--collect-all", "torch")
+}
 
 # 2. Grab a static ffmpeg build (BtbN build — includes libmp3lame)
 $ffdir = "third_party\ffmpeg\bin"
@@ -33,6 +48,7 @@ if (-not (Test-Path "$ffdir\ffmpeg.exe")) {
     --add-binary "$ffdir\ffprobe.exe;." `
     --add-data "assets\icon.png;assets" `
     --icon "assets\icon.ico" `
+    @AI_FLAGS `
     voicegrab.py
 
 Write-Host "Built dist\VoiceGrab.exe"

requirements-ai.txt

@@ -0,0 +1,19 @@
+# VoiceGrab — OPTIONAL AI voice separation (heavy mode)
+#
+# The core app does NOT need this file; it only enables the
+# "AI voice separation (Demucs)" mode.
+#
+# Install:
+#   pip install -r requirements-ai.txt
+#
+# Notes:
+# - `torch` from PyPI on Windows/Linux ships CUDA wheels (~2 GB with the
+#   nvidia-* deps). CPU-only machines can install the smaller CPU build:
+#     pip install torch --index-url https://download.pytorch.org/whl/cpu
+# - The Demucs `htdemucs_ft` model (~90 MB) downloads on first use and is
+#   cached under the app data dir (<appdata>/VoiceGrab/models/).
+# - Licenses: see THIRD_PARTY.md (code MIT; weights usable per Meta's
+#   model license — fine for personal use).
+
+torch>=2.4,<3
+demucs>=4.1,<4.2

tests/test_separators.py

@@ -0,0 +1,252 @@
+#!/usr/bin/env python3
+"""VoiceGrab separator tests (GPU-upgrade.md §6 automation).
+
+Run with the project venv:
+    .venv/bin/python tests/test_separators.py
+or with pytest:
+    .venv/bin/pytest tests/
+
+The AI (Demucs) end-to-end test needs the model cached or a network
+connection (~90 MB, first run only) and takes ~15 s on CPU.
+"""
+from __future__ import annotations
+
+import math
+import os
+import subprocess
+import sys
+import tempfile
+import wave
+
+HERE = os.path.dirname(os.path.abspath(__file__))
+ROOT = os.path.dirname(HERE)
+sys.path.insert(0, ROOT)
+
+import numpy as np  # noqa: E402
+
+import voicegrab as vg  # noqa: E402
+
+
+# ---------------------------------------------------------------------------
+# fixtures / helpers
+# ---------------------------------------------------------------------------
+
+def _synthetic_voice(n: int, sr: int = 48000) -> np.ndarray:
+    """Speech-like signal (130 Hz fundamental + harmonics, syllable AM)."""
+    t = np.arange(n) / sr
+    sig = np.zeros_like(t)
+    for h in range(1, 31):
+        sig += (0.6 / h) * np.sin(2 * np.pi * 130.0 * h * t + h * 0.7)
+    sig *= 0.35 + 0.65 * (0.5 + 0.5 * np.sin(2 * np.pi * 4.5 * t))
+    sig /= np.abs(sig).max()
+    rng = np.random.default_rng(7)
+    noise = 0.5 * np.cumsum(rng.normal(0, 1, len(t)))
+    noise /= np.abs(noise).max()
+    return np.clip(0.7 * sig + 0.5 * noise, -1, 1).astype(np.float32)
+
+
+def _write_test_mp3(path: str, seconds: int = 5) -> str:
+    """Create a seconds-long mp3 with ffmpeg (voice-like tone + noise)."""
+    cmd = [
+        "ffmpeg", "-hide_banner", "-v", "error", "-y",
+        "-f", "lavfi", "-i", f"sine=frequency=220:duration={seconds}",
+        "-f", "lavfi", "-i", f"anoisesrc=d={seconds}:c=pink:a=0.3",
+        "-filter_complex",
+        "[0][1]amix=inputs=2:weights=1 0.5,atrim=0:" + str(seconds) + ",asetpts=PTS-STARTPTS[out]",
+        "-map", "[out]", "-c:a", "libmp3lame", path,
+    ]
+    subprocess.run(cmd, check=True, capture_output=True)
+    return path
+
+
+def _mp3_duration(path: str) -> float:
+    out = subprocess.run(
+        ["ffprobe", "-v", "error", "-show_entries", "format=duration",
+         "-of", "default=nw=1:nk=1", path],
+        capture_output=True, text=True, check=True).stdout.strip()
+    return float(out)
+
+
+def _mp3_bitrate(path: str) -> int:
+    out = subprocess.run(
+        ["ffprobe", "-v", "error", "-show_entries", "format=bit_rate",
+         "-of", "default=nw=1:nk=1", path],
+        capture_output=True, text=True, check=True).stdout.strip()
+    return int(out)
+
+
+def _ai_available() -> bool:
+    return vg._ai_deps_present()
+
+
+# ---------------------------------------------------------------------------
+# unit tests
+# ---------------------------------------------------------------------------
+
+def test_modes_constants():
+    assert vg.MODE_NONE == "none"
+    assert vg.MODE_DSP == "dsp"
+    assert vg.MODE_AI == "ai"
+    assert vg.AI_MODEL_NAME == "htdemucs_ft"
+
+
+def test_dsp_available_and_shape():
+    sep = vg.DSSeparator(intensity=0.8, stationary=True)
+    ok, reason = sep.available()
+    assert ok, f"DSP should be available in this venv: {reason}"
+    y = _synthetic_voice(48000)
+    out, sr = sep.separate(y, 48000, lambda m: None)
+    assert sr == 48000, "DSP must not change the sample rate"
+    assert out.dtype == np.float32
+    assert out.ndim == 1 and len(out) == len(y)
+    assert np.isfinite(out).all(), "no NaN/Inf allowed"
+    assert float(np.sqrt(np.mean(out**2))) > 0, "must not be all-silence"
+
+
+def test_ai_available_report():
+    sep = vg.DemucsSeparator()
+    ok, reason = sep.available()
+    if _ai_available():
+        assert ok
+    else:
+        assert not ok
+        assert "requirements-ai.txt" in reason
+
+
+def test_ai_graceful_without_torch():
+    """DemucsSeparator.available() must degrade gracefully when torch is absent."""
+    code = (
+        "import sys, types\n"
+        "class Blocker:\n"
+        "    def find_spec(self, name, path=None, target=None):\n"
+        "        if name == 'torch' or name.startswith('torch.'):\n"
+        "            raise ModuleNotFoundError('blocked: ' + name)\n"
+        "        return None\n"
+        "sys.meta_path.insert(0, Blocker())\n"
+        "sys.path.insert(0, %r)\n"
+        "import voicegrab as vg\n"
+        "ok, reason = vg.DemucsSeparator().available()\n"
+        "assert not ok, 'should report unavailable when torch is blocked'\n"
+        "assert 'requirements-ai.txt' in reason\n"
+        "print('graceful-degradation OK')\n"
+    ) % ROOT
+    r = subprocess.run([sys.executable, "-c", code],
+                       capture_output=True, text=True, timeout=120)
+    assert r.returncode == 0, r.stderr
+    assert "graceful-degradation OK" in r.stdout
+
+
+def test_ai_pick_device_order():
+    if not _ai_available():
+        print("  (skipped: torch/demucs not installed)")
+        return
+    sep = vg.DemucsSeparator(device="cuda")
+    assert sep._pick_device() == "cuda", "explicit device must be honored"
+    auto = vg.DemucsSeparator()._pick_device()
+    assert auto in ("cpu", "cuda", "mps")
+
+
+def test_ai_separate_unit():
+    if not _ai_available():
+        print("  (skipped: torch/demucs not installed)")
+        return
+    sep = vg.DemucsSeparator()
+    y = _synthetic_voice(3 * 48000)   # 3 s @ 48 kHz
+    out, sr = sep.separate(y, 48000, lambda m: None)
+    assert sr == 44100, "Demucs runs at 44.1 kHz"
+    assert out.dtype == np.float32 and out.ndim == 1
+    assert np.isfinite(out).all()
+    expected = 3 * 44100
+    assert abs(len(out) - expected) <= 44100 * 0.02, f"duration drift: {len(out)} vs {expected}"
+    assert float(np.sqrt(np.mean(out**2))) > 0, "must not be all-silence"
+
+
+# ---------------------------------------------------------------------------
+# regression: full CLI pipeline (GPU-upgrade.md §6.2)
+# ---------------------------------------------------------------------------
+
+def test_cli_dsp_regression():
+    with tempfile.TemporaryDirectory(prefix="voicegrab-test-") as wd:
+        src = os.path.join(wd, "in.mp3")
+        out = os.path.join(wd, "out.mp3")
+        _write_test_mp3(src, 5)
+        r = subprocess.run(
+            [sys.executable, os.path.join(ROOT, "voicegrab.py"),
+             "--cli", src, "0", "5", out, "--mode", "dsp"],
+            capture_output=True, text=True, timeout=300)
+        assert r.returncode == 0, r.stderr
+        assert os.path.exists(out)
+        dur = _mp3_duration(out)
+        assert abs(dur - 5.0) <= 0.15, f"duration {dur} not within 5.0 ± 0.15 s"
+        br = _mp3_bitrate(out)
+        assert 300_000 <= br <= 350_000, f"expected ~320 kbps, got {br}"
+
+
+def test_cli_ai_regression():
+    if not _ai_available():
+        print("  (skipped: torch/demucs not installed)")
+        return
+    with tempfile.TemporaryDirectory(prefix="voicegrab-test-") as wd:
+        src = os.path.join(wd, "in.mp3")
+        out = os.path.join(wd, "out.mp3")
+        _write_test_mp3(src, 5)
+        r = subprocess.run(
+            [sys.executable, os.path.join(ROOT, "voicegrab.py"),
+             "--cli", src, "0", "5", out, "--mode", "ai"],
+            capture_output=True, text=True, timeout=600)
+        assert r.returncode == 0, r.stderr
+        assert os.path.exists(out)
+        dur = _mp3_duration(out)
+        assert abs(dur - 5.0) <= 0.15, f"duration {dur} not within 5.0 ± 0.15 s"
+        br = _mp3_bitrate(out)
+        assert 300_000 <= br <= 350_000, f"expected ~320 kbps, got {br}"
+
+
+def test_cli_none_mode():
+    with tempfile.TemporaryDirectory(prefix="voicegrab-test-") as wd:
+        src = os.path.join(wd, "in.mp3")
+        out = os.path.join(wd, "out.mp3")
+        _write_test_mp3(src, 5)
+        r = subprocess.run(
+            [sys.executable, os.path.join(ROOT, "voicegrab.py"),
+             "--cli", src, "0", "5", out, "--mode", "none"],
+            capture_output=True, text=True, timeout=300)
+        assert r.returncode == 0, r.stderr
+        assert abs(_mp3_duration(out) - 5.0) <= 0.15
+
+
+def test_cli_rejects_bad_mode():
+    with tempfile.TemporaryDirectory(prefix="voicegrab-test-") as wd:
+        src = os.path.join(wd, "in.mp3")
+        _write_test_mp3(src, 5)
+        r = subprocess.run(
+            [sys.executable, os.path.join(ROOT, "voicegrab.py"),
+             "--cli", src, "0", "5", os.path.join(wd, "o.mp3"), "--mode", "bogus"],
+            capture_output=True, text=True, timeout=120)
+        assert r.returncode == 2
+        assert "none | dsp | ai" in r.stdout
+
+
+# ---------------------------------------------------------------------------
+
+def main() -> int:
+    tests = [v for k, v in sorted(globals().items()) if k.startswith("test_")]
+    failed = 0
+    for t in tests:
+        name = t.__name__
+        try:
+            t()
+            print(f"PASS  {name}")
+        except AssertionError as exc:
+            failed += 1
+            print(f"FAIL  {name}: {exc}")
+        except Exception as exc:  # noqa: BLE001
+            failed += 1
+            print(f"ERROR {name}: {type(exc).__name__}: {exc}")
+    total = len(tests)
+    print(f"\n{total - failed}/{total} passed")
+    return 1 if failed else 0
+
+
+if __name__ == "__main__":
+    raise SystemExit(main())

voicegrab.py

@@ -3,22 +3,27 @@
 
 - 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)
+- Optional voice processing (one of, mutually exclusive):
+    * fast noise reduction (CPU DSP via noisereduce — no downloads)
+    * AI voice separation (Demucs htdemucs_ft — GPU if available, optional install)
+    * no processing (loudness-normalize only)
 - 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 math
 import os
 import platform
 import shutil
 import subprocess
 import sys
 import tempfile
+import threading
 import wave
 
 APP_NAME = "VoiceGrab"
-APP_VERSION = "1.0.0"
+APP_VERSION = "1.1.0"
 
 # ---------------------------------------------------------------------------
 # Resource / executable helpers
@@ -33,7 +38,7 @@ def app_dir() -> str:
 
 
 def appdata_dir() -> str:
-    """Per-user directory for saved model noise profiles."""
+    """Per-user directory for cached AI models (and any saved profiles)."""
     if platform.system() == "Windows":
         base = os.environ.get("APPDATA", os.path.expanduser("~"))
     elif platform.system() == "Darwin":
@@ -179,6 +184,203 @@ class VoiceReducer:
         return nr.reduce_noise(**kwargs)
 
 
+# ---------------------------------------------------------------------------
+# Separation backends — one mode at a time (DSP and AI are alternatives, not
+# stacked: running noisereduce after Demucs adds double-processing artifacts)
+# ---------------------------------------------------------------------------
+
+MODE_NONE = "none"   # loudness-normalize only
+MODE_DSP = "dsp"     # fast STFT noise reduction (noisereduce)
+MODE_AI = "ai"       # deep-learning separation (Demucs htdemucs_ft)
+
+AI_MODEL_NAME = "htdemucs_ft"   # MIT-licensed weights; we keep its "vocals" stem
+
+
+class Separator:
+    """Common interface for the voice-processing modes."""
+
+    name = "base"
+
+    def available(self) -> tuple[bool, str]:
+        """Cheap check (no heavy imports). Returns (ok, human-readable reason)."""
+        raise NotImplementedError
+
+    def separate(self, y: "np.ndarray", sr: int, log=print) -> tuple["np.ndarray", int]:
+        """Return (float32 mono audio, sample_rate)."""
+        raise NotImplementedError
+
+    def cancel(self):
+        """Best-effort cancel of a running separation (no-op for fast modes)."""
+
+
+def _ai_deps_present() -> bool:
+    """Fast presence check for torch+demucs without importing them (importing
+    torch takes ~1-2 s and ~400 MB RAM — too expensive for a startup UI check)."""
+    import importlib.util
+    try:
+        return (importlib.util.find_spec("torch") is not None
+                and importlib.util.find_spec("demucs") is not None)
+    except Exception:
+        return False
+
+
+class DSSeparator(Separator):
+    """Current noisereduce STFT path — unchanged behavior."""
+
+    name = MODE_DSP
+
+    def __init__(self, intensity: float = 0.8, stationary: bool = False):
+        self.intensity = float(intensity)
+        self.stationary = stationary
+
+    def available(self) -> tuple[bool, str]:
+        if not HAVE_NR:
+            return False, (f"noisereduce is not installed "
+                           f"({NR_IMPORT_ERROR or 'unknown import error'})")
+        return True, ""
+
+    def separate(self, y, sr, log=print) -> tuple["np.ndarray", int]:
+        log("Isolating voice — fast noise reduction (CPU)…")
+        red = VoiceReducer(self.intensity, self.stationary)
+        return red.reduce(y, sr), sr
+
+
+class DemucsSeparator(Separator):
+    """AI voice separation with Demucs (htdemucs_ft). Lazy imports; GPU-aware.
+
+    Only imported/used when the AI mode is actually selected, so the core app
+    stays a hard-dependency-free ~100 MB tool.
+    """
+
+    name = MODE_AI
+
+    def __init__(self, model_name: str = AI_MODEL_NAME, device: str | None = None):
+        self.model_name = model_name
+        self.device = device          # None → auto-pick: cuda → mps → cpu
+        self._api = None              # demucs.api.Separator (lazy, worker thread)
+        self._device_used = None
+        self._cancel = threading.Event()
+
+    # -- cheap checks (safe to call from the GUI thread) -------------------
+    def available(self) -> tuple[bool, str]:
+        if not _ai_deps_present():
+            return False, ("demucs/torch not installed — "
+                           "pip install -r requirements-ai.txt "
+                           "(pulls torch, ~2 GB with CUDA wheels)")
+        return True, ""
+
+    def cancel(self):
+        self._cancel.set()
+
+    # -- heavy work (worker thread only) ------------------------------------
+    def _pick_device(self) -> str:
+        if self.device:
+            return self.device
+        import torch
+        if torch.cuda.is_available():
+            return "cuda"
+        mps = getattr(torch.backends, "mps", None)
+        if mps is not None and mps.is_available():
+            return "mps"
+        return "cpu"
+
+    def _get_api(self, log):
+        if self._api is None:
+            # Cache downloads under <appdata>/VoiceGrab/models. Must be set
+            # before torch/huggingface_hub are first imported in this thread.
+            root = os.path.join(appdata_dir(), "models")
+            os.makedirs(root, exist_ok=True)
+            os.environ.setdefault("TORCH_HOME", os.path.join(root, "torch"))
+            os.environ.setdefault("HF_HOME", os.path.join(root, "huggingface"))
+            from demucs.api import Separator as _DemucsApi
+            self._device_used = self._pick_device()
+            log(f"Loading AI model '{self.model_name}' "
+                f"(first use downloads ~90 MB, cached afterwards)…")
+            self._api = _DemucsApi(self.model_name, device=self._device_used)
+            log("AI model ready.")
+        return self._api
+
+    @staticmethod
+    def _model_segment(model) -> float:
+        try:
+            for sub in model.models:      # BagOfModels
+                seg = getattr(sub, "segment", None)
+                if seg:
+                    return float(seg)
+        except AttributeError:
+            pass
+        seg = getattr(model, "segment", None)
+        return float(seg) if seg else 8.0
+
+    @staticmethod
+    def _estimate_total_chunks(d: dict, api) -> int:
+        """Expected 'end' callback count = submodels × shifts × ceil(len/stride)."""
+        try:
+            length = int(d["audio_length"])
+            seg = DemucsSeparator._model_segment(api.model)
+            n_subs = int(d.get("models", 1) or 1)
+            stride = int(0.75 * seg * api.samplerate)   # overlap=0.25
+            if stride <= 0:
+                return 0
+            return max(1, math.ceil(length / stride)) * n_subs
+        except Exception:
+            return 0   # unknown → log raw chunk counts instead
+
+    def separate(self, y, sr, log=print) -> tuple["np.ndarray", int]:
+        import numpy as np
+        import torch
+
+        api = self._get_api(log)
+        device = self._device_used
+        if device == "cuda":
+            log("AI voice separation on GPU (CUDA)…")
+        elif device == "mps":
+            log("AI voice separation on GPU (Apple MPS)…")
+        else:
+            log("AI voice separation on CPU (no GPU detected — expect a wait)…")
+
+        # (C, T) float32 at the original rate; demucs resamples to 44.1 kHz
+        # and duplicates mono→stereo for us (see demucs.audio.convert_audio).
+        wav = torch.from_numpy(np.ascontiguousarray(y, dtype=np.float32))
+        if wav.ndim == 1:
+            wav = wav[None, :]
+
+        state = {"done": 0, "total": None, "last_pct": -1}
+
+        def _progress(d):
+            if self._cancel.is_set():
+                raise KeyboardInterrupt()   # demucs' documented way to abort
+            if state["total"] is None:
+                state["total"] = self._estimate_total_chunks(d, api)
+            if d.get("state") == "end":
+                state["done"] += 1
+                total = state["total"]
+                if total:
+                    pct = min(99, int(state["done"] * 100 / total))
+                    if state["done"] >= total or pct >= state["last_pct"] + 10:
+                        state["last_pct"] = pct
+                        log(f"AI separation: {state['done']}/{total} chunks ({pct}%)…")
+                else:
+                    log(f"AI separation: chunk {state['done']} done…")
+
+        self._cancel.clear()
+        try:
+            api.update_parameter(callback=_progress)
+            _ref, stems = api.separate_tensor(wav, sr)
+        except KeyboardInterrupt:
+            raise FfmpegError("Cancelled by user.")
+
+        key = ("vocals" if "vocals" in stems
+               else "speech" if "speech" in stems
+               else next(iter(stems)))
+        v = stems[key]
+        while v.dim() > 1:
+            v = v[0]                       # first channel → mono
+        out = v.detach().cpu().numpy().astype(np.float32)
+        out = np.where(np.isfinite(out), out, 0.0)   # defensive: never emit NaN/Inf
+        return out, int(api.samplerate)
+
+
 # ---------------------------------------------------------------------------
 # Audio IO helpers (numpy/soundfile)
 # ---------------------------------------------------------------------------
@@ -199,21 +401,24 @@ def write_wav_f32(path: str, y, sr: int):
 # GUI
 # ---------------------------------------------------------------------------
 
+
 def _import_pyside():
     from PySide6 import QtCore, QtGui, QtWidgets  # noqa: F401
     return QtCore, QtGui, QtWidgets
 
 
-def run_gui() -> int:
+def _peak_normalize(y):
-    QtCore, QtGui, QtWidgets = _import_pyside()
+    """Peak-normalize (99.5th percentile) to ~ -1 dBFS reference."""
-
-    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
 
+
+def run_gui() -> int:
+    QtCore, QtGui, QtWidgets = _import_pyside()
+
     class Worker(QtCore.QThread):
         log = QtCore.Signal(str)
         done = QtCore.Signal(object, str)   # (success, message)
@@ -221,6 +426,11 @@ def run_gui() -> int:
         def __init__(self, job):
             super().__init__()
             self.job = job
+            self._sep: Separator | None = None
+
+        def cancel(self):
+            if self._sep is not None:
+                self._sep.cancel()
 
         def run(self):
             import numpy as _np
@@ -230,15 +440,26 @@ def run_gui() -> int:
                     wav = extract_wav(self.job["input"], self.job["start"], self.job["stop"], wd)
                     y, sr = read_wav_f32(wav)
 
-                    if self.job["isolate"]:
+                    mode = self.job.get("mode", MODE_DSP)
-                        if not HAVE_NR:
+                    if mode == MODE_AI:
-                            raise FfmpegError(f"Voice isolation unavailable: {NR_IMPORT_ERROR}")
+                        self._sep = DemucsSeparator()
-                        self.log.emit("Isolating voice (noise reduction)\u2026")
+                    elif mode == MODE_DSP:
-                        red = VoiceReducer(self.job["intensity"], self.job["stationary"])
+                        self._sep = DSSeparator(self.job["intensity"], self.job["stationary"])
-                        y = red.reduce(y, sr)
+                    else:
+                        self._sep = None
+                    if self._sep is not None:
+                        ok, reason = self._sep.available()
+                        if not ok:
+                            raise FfmpegError(f"Voice processing unavailable: {reason}")
+                        y, sr = self._sep.separate(y, sr, self.log.emit)
+                        if mode == MODE_AI:
+                            # Separated vocals sit a few dB below the mix; bring
+                            # peaks up before loudnorm (post-gain step).
+                            self.log.emit("Leveling separated vocals…")
+                            y = _peak_normalize(y)
                     else:
                         self.log.emit("Normalizing loudness…")
-                        y = _normalize(y)
+                        y = _peak_normalize(y)
 
                     work_wav = os.path.join(wd, "work.wav")
                     write_wav_f32(work_wav, y, sr)
@@ -257,10 +478,10 @@ def run_gui() -> int:
                     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)
+    return _main_loop(QtCore, QtGui, QtWidgets, Worker)
 
 
-def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
+def _main_loop(QtCore, QtGui, QtWidgets, Worker) -> int:
     app = QtWidgets.QApplication(sys.argv)
     app.setApplicationName(APP_NAME)
     app.setApplicationVersion(APP_VERSION)
@@ -321,11 +542,26 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
             row2.addWidget(btn_listen)
             lay.addLayout(row2)
 
-            # Row 3: isolation + output
+            # Row 3: processing mode (one of: fast DSP / AI / none)
-            self.chk_isolate = QtWidgets.QCheckBox("Isolate voice (reduce background noise)")
+            proc_box = QtWidgets.QGroupBox("Voice processing (choose one)")
-            self.chk_isolate.setChecked(True)
+            proc_lay = QtWidgets.QVBoxLayout(proc_box)
-            self.chk_isolate.toggled.connect(self._isolate_toggled)
+            self.rb_dsp = QtWidgets.QRadioButton(
-            lay.addWidget(self.chk_isolate)
+                "Fast noise reduction (CPU) — fan / hum / room tone; no downloads")
+            self.rb_ai = QtWidgets.QRadioButton(
+                "AI voice separation (Demucs) — background music & second speakers; "
+                "uses GPU if available")
+            self.rb_none = QtWidgets.QRadioButton(
+                "No processing — just loudness-normalize")
+            self.rb_dsp.setChecked(True)
+            self.rb_ai.setToolTip(
+                "Deep-learning vocal separation. First use downloads ~90 MB of models. "
+                "GPU (NVIDIA CUDA / Apple) is strongly recommended; on CPU a 60 s clip "
+                "takes a few minutes. Long AI runs can be cancelled between chunks.")
+            for rb in (self.rb_dsp, self.rb_ai, self.rb_none):
+                proc_lay.addWidget(rb)
+            lay.addWidget(proc_box)
+            self.rb_dsp.toggled.connect(self._proc_mode_changed)
+            self.rb_ai.toggled.connect(self._proc_mode_changed)
 
             iso_row = QtWidgets.QHBoxLayout()
             iso_row.addWidget(QtWidgets.QLabel("Noise reduction strength:"))
@@ -339,11 +575,12 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
                 "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
+
+            self.lbl_proc_status = QtWidgets.QLabel("")
+            self.lbl_proc_status.setStyleSheet("color:#64748b;")
+            self.lbl_proc_status.setWordWrap(True)
+            lay.addWidget(self.lbl_proc_status)
 
             out_row = QtWidgets.QHBoxLayout()
             out_row.addWidget(QtWidgets.QLabel("Output MP3:"))
@@ -363,6 +600,14 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
 
             # Bottom bar
             bar = QtWidgets.QHBoxLayout()
+            self.btn_cancel = QtWidgets.QPushButton("Cancel")
+            self.btn_cancel.setMinimumHeight(40)
+            self.btn_cancel.setEnabled(False)
+            self.btn_cancel.setToolTip(
+                "Cancels an in-progress AI separation at the next chunk boundary. "
+                "Fast DSP runs finish in seconds and are not cancelable.")
+            self.btn_cancel.clicked.connect(self._cancel)
+            bar.addWidget(self.btn_cancel)
             self.btn_export = QtWidgets.QPushButton("Export MP3")
             self.btn_export.setMinimumHeight(40)
             self.btn_export.setStyleSheet(
@@ -375,15 +620,32 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
             bar.addWidget(self.lbl_status, 1)
             lay.addLayout(bar)
 
-            if not HAVE_NR:
+            # ----- availability / default mode (cheap checks only) -----
-                self.lbl_iso_status.setText(f"⚠ voice isolation unavailable ({NR_IMPORT_ERROR})")
+            dsp_ok, dsp_reason = DSSeparator().available()
-                self.chk_isolate.setEnabled(False)
+            ai_ok, ai_reason = DemucsSeparator().available()
+            self.spn_intensity.setEnabled(dsp_ok)
+            self.chk_stationary.setEnabled(dsp_ok)
+            if not dsp_ok:
+                self.rb_dsp.setEnabled(False)
+                self.rb_none.setChecked(True)
+            self.rb_ai.setEnabled(ai_ok)
+            self._proc_mode_changed(self.rb_dsp.isChecked())
+            hints = []
+            if not ai_ok:
+                hints.append(f"⚠ AI separation unavailable — {ai_reason}")
+            else:
+                hints.append(
+                    "AI mode ready — GPU (NVIDIA/Apple) used if present, otherwise CPU "
+                    "(slower); first use downloads ~90 MB of models, cached afterwards.")
+            if not dsp_ok:
+                hints.append(f"⚠ fast noise reduction unavailable — {dsp_reason}")
+            self.lbl_proc_status.setText(" ".join(hints))
 
         # ---------- helpers ----------
-        def _isolate_toggled(self, on):
+        def _proc_mode_changed(self, *_):
-            for w in self.iso_row_widget.items():
+            dsp = self.rb_dsp.isChecked()
-                if isinstance(w, QtWidgets.QWidget) and w not in (self.lbl_iso_status,):
+            self.spn_intensity.setEnabled(dsp)
-                    w.setEnabled(on)
+            self.chk_stationary.setEnabled(dsp)
 
         @staticmethod
         def _parse_time(s: str) -> float:
@@ -515,11 +777,18 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
                 out = os.path.join(os.path.dirname(os.path.abspath(self.input_path)),
                                    base + "_voicegrab.mp3")
             self.btn_export.setEnabled(False)
+            self.btn_cancel.setEnabled(True)
             self.status("Working… (see log in status area)")
+            if self.rb_ai.isChecked():
+                mode = MODE_AI
+            elif self.rb_none.isChecked():
+                mode = MODE_NONE
+            else:
+                mode = MODE_DSP
             self.worker = Worker({
                 "input": self.input_path, "start": s, "stop": e,
                 "output": out,
-                "isolate": self.chk_isolate.isChecked(),
+                "mode": mode,
                 "intensity": self.spn_intensity.value() / 100.0,
                 "stationary": self.chk_stationary.isChecked(),
                 "lufts": -16,
@@ -528,8 +797,15 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
             self.worker.done.connect(self._export_done)
             self.worker.start()
 
+        def _cancel(self):
+            if self.worker is None:
+                return
+            self.worker.cancel()
+            self.status("Cancelling… (AI separation stops at the next chunk boundary)")
+
         def _export_done(self, ok, msg):
             self.btn_export.setEnabled(True)
+            self.btn_cancel.setEnabled(False)
             self.worker = None
             if ok:
                 self.status(f"✔ Saved: {msg}")
@@ -580,10 +856,23 @@ def _main_loop(QtCore, QtGui, QtWidgets, Worker, _normalize) -> int:
 
 if __name__ == "__main__":
     if "--cli" in sys.argv:
-        # Simple headless mode: voicegrab --cli <input> <start> <stop> [output.mp3]
+        # Simple headless mode:
+        #   voicegrab --cli <input> <start> <stop> [output.mp3] [--mode none|dsp|ai]
         sys.argv = [a for a in sys.argv if a != "--cli"]
+        mode = MODE_DSP
+        if "--mode" in sys.argv:
+            i = sys.argv.index("--mode")
+            try:
+                mode = sys.argv[i + 1]
+            except IndexError:
+                print("usage: --mode needs a value: none | dsp | ai")
+                raise SystemExit(2)
+            del sys.argv[i:i + 2]
+        if mode not in (MODE_NONE, MODE_DSP, MODE_AI):
+            print(f"unknown --mode {mode!r} (use: none | dsp | ai)")
+            raise SystemExit(2)
         if len(sys.argv) < 4:
-            print("usage: voicegrab --cli <input> <start> <stop> [output.mp3]")
+            print("usage: voicegrab --cli <input> <start> <stop> [output.mp3] [--mode none|dsp|ai]")
             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(
@@ -593,9 +882,21 @@ if __name__ == "__main__":
         with tempfile.TemporaryDirectory(prefix="voicegrab-") as wd:
             wav = extract_wav(inp, s, e, wd)
             y, sr = read_wav_f32(wav)
-            if HAVE_NR:
+            if mode == MODE_AI:
-                red = VoiceReducer(0.8, stationary=True)
+                sep = DemucsSeparator()
-                y = red.reduce(y, sr)
+                ok, reason = sep.available()
+                if not ok:
+                    print(f"AI separation unavailable: {reason}")
+                    raise SystemExit(3)
+                y, sr = sep.separate(y, sr, print)
+                y = _peak_normalize(y)   # post-gain before loudnorm
+            elif mode == MODE_DSP:
+                sep = DSSeparator(intensity=0.8, stationary=True)
+                ok, reason = sep.available()
+                if not ok:
+                    print(f"Fast noise reduction unavailable: {reason}")
+                    raise SystemExit(3)
+                y, sr = sep.separate(y, sr, print)
             red_wav = os.path.join(wd, "reduced.wav")
             write_wav_f32(red_wav, y, sr)
             subprocess.run([