fix(backend): pin miniaudio in requirements-mlx.txt (#505)

mlx-audio's STT path imports miniaudio, but we install mlx-audio
--no-deps to dodge its transformers>=5.x pin. Nothing else pulls
miniaudio transitively, so fresh Apple Silicon installs fail to
transcribe with ModuleNotFoundError: miniaudio. Listed explicitly
and updated the stale comments in requirements-mlx.txt and
release.yml that claimed it came from other engines.

Co-Authored-By: Claude Opus 4.7 (1M context) <[email protected]>

.github/workflows/release.yml

@@ -26,12 +26,45 @@ jobs:
             args: ""
             python-version: "3.12"
             backend: "pytorch"
+          - platform: "ubuntu-22.04"
+            # --config override disables updater-artifact generation on Linux.
+            # tauri.conf.json has createUpdaterArtifacts: "v1Compatible" which
+            # on Linux wants to synthesize a .AppImage.tar.gz by downloading
+            # linuxdeploy at build time — this is what silently hangs CI
+            # (see v0.4.2 round 2, 25 min of no output after rpm bundling).
+            # We ship deb+rpm only; Linux users update via apt/dnf, not the
+            # Tauri in-app updater.
+            args: '--target x86_64-unknown-linux-gnu --bundles deb,rpm --verbose --config {"bundle":{"createUpdaterArtifacts":false}}'
+            python-version: "3.12"
+            backend: "pytorch"
 
     runs-on: ${{ matrix.platform }}
 
     steps:
       - uses: actions/checkout@v4
 
+      # Ubuntu runners ship with ~14 GB free; pip + PyInstaller + torch can
+      # peak well above that during the build. Reclaim ~25 GB by pruning
+      # preinstalled toolchains we don't use. This is what likely tripped
+      # the March 2026 Linux release attempts (see commit 103e98b
+      # "github runners suck") — not a code issue, a disk-pressure one.
+      - name: Free up disk space (ubuntu)
+        if: contains(matrix.platform, 'ubuntu') || contains(matrix.platform, 'namespace')
+        # Pinned to v1.3.1 (SHA) — this job runs with contents: write and
+        # handles signing secrets later, so we don't want a floating ref.
+        uses: jlumbroso/free-disk-space@54081f138730dfa15788a46383842cd2f914a1be
+        with:
+          tool-cache: false
+          android: true
+          dotnet: true
+          haskell: true
+          # large-packages: true would `apt-get remove '^llvm-.*'`, which
+          # cascade-removes reverse deps that won't be pulled back in by the
+          # `llvm-dev` install below. The other flags already free ~20 GB,
+          # enough for the Python + torch + PyInstaller build.
+          large-packages: false
+          swap-storage: true
+
       - name: Install dependencies (ubuntu only)
         if: contains(matrix.platform, 'ubuntu') || contains(matrix.platform, 'namespace')
         run: |
@@ -73,8 +106,10 @@ jobs:
           # fine on transformers 4.57.x in practice (verified in dev), so install
           # them --no-deps. mlx-audio's other runtime deps (huggingface_hub,
           # librosa, numpy, numba, pyloudnorm) are already in requirements.txt;
-          # the rest (sounddevice, miniaudio, protobuf, sentencepiece, pyyaml,
-          # jinja2) are pulled in by other engines.
+          # miniaudio is in requirements-mlx.txt (needed by mlx_audio.stt,
+          # not transitively pulled by anything else — see issue #505); the
+          # rest (sounddevice, protobuf, sentencepiece, pyyaml, jinja2) are
+          # pulled in by other engines.
           pip install --no-deps mlx-lm==0.31.1
           pip install --no-deps mlx-audio==0.4.1
 
@@ -133,6 +168,21 @@ jobs:
           p12-file-base64: ${{ secrets.APPLE_CERTIFICATE }}
           p12-password: ${{ secrets.APPLE_CERTIFICATE_PASSWORD }}
 
+      - name: Disk / environment snapshot (pre-bundle debug)
+        if: contains(matrix.platform, 'ubuntu') || contains(matrix.platform, 'namespace')
+        run: |
+          echo "=== df -h ==="
+          df -h
+          echo "=== free -h ==="
+          free -h
+          echo "=== Rust / Cargo ==="
+          rustc --version
+          cargo --version
+          echo "=== Bun ==="
+          bun --version
+          echo "=== Tauri CLI ==="
+          cd tauri && bun run tauri --version
+
       - name: Extract release notes from CHANGELOG.md
         id: changelog
         shell: bash
@@ -156,7 +206,13 @@ jobs:
             echo "CHANGELOG_EOF"
           } >> "$GITHUB_OUTPUT"
 
+      # Linux hang watchdog: previous releases silently wedged inside tauri
+      # bundling (possibly linuxdeploy/AppImage download, possibly cargo link).
+      # Cap the step at 30 min so we get logs instead of waiting out the 6hr
+      # job timeout. Other platforms historically complete in ~25 min, so 45
+      # is comfortable.
       - uses: tauri-apps/[email protected]
+        timeout-minutes: ${{ (contains(matrix.platform, 'ubuntu') || contains(matrix.platform, 'namespace')) && 30 || 45 }}
         env:
           GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
           TAURI_SIGNING_PRIVATE_KEY: ${{ secrets.TAURI_SIGNING_PRIVATE_KEY }}
@@ -168,6 +224,9 @@ jobs:
           APPLE_PROVIDER_SHORT_NAME: ${{ secrets.APPLE_PROVIDER_SHORT_NAME }}
           APPLE_API_ISSUER: ${{ secrets.APPLE_API_ISSUER }}
           APPLE_API_KEY: ${{ secrets.APPLE_API_KEY }}
+          # Stream subprocess stdout/stderr so the hang is visible in logs.
+          CARGO_TERM_VERBOSE: "true"
+          RUST_BACKTRACE: "1"
         with:
           projectPath: tauri
           tagName: v__VERSION__

app/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@voicebox/app",
-  "version": "0.4.1",
+  "version": "0.4.2",
   "private": true,
   "type": "module",
   "scripts": {

app/src/components/AppFrame/AppFrame.tsx

@@ -1,5 +1,6 @@
 import { useRouterState } from '@tanstack/react-router';
 import { TitleBarDragRegion } from '@/components/TitleBarDragRegion';
+import { AudioKeepAlive } from '@/components/AudioPlayer/AudioKeepAlive';
 import { AudioPlayer } from '@/components/AudioPlayer/AudioPlayer';
 import { StoryTrackEditor } from '@/components/StoriesTab/StoryTrackEditor';
 import { TOP_SAFE_AREA_PADDING } from '@/lib/constants/ui';
@@ -26,6 +27,7 @@ export function AppFrame({ children }: AppFrameProps) {
       className={cn('h-screen bg-background flex flex-col overflow-hidden', TOP_SAFE_AREA_PADDING)}
     >
       <TitleBarDragRegion />
+      <AudioKeepAlive />
       {children}
       {showTrackEditor ? (
         <StoryTrackEditor storyId={story.id} items={story.items} />

app/src/components/AudioPlayer/AudioKeepAlive.tsx

@@ -0,0 +1,85 @@
+import { useEffect, useRef } from 'react';
+import { debug } from '@/lib/utils/debug';
+
+// WKWebView tears down the app's CoreAudio output when idle for long enough,
+// and a JS-level reload (cmd+R) does NOT restore it — only relaunching the
+// Tauri app does. Keeping a silent <audio> element looping forever prevents
+// the OS audio session from ever going dormant.
+//
+// Real silence (zero PCM samples) at full volume is preferred over a muted
+// element: browsers/WebKit can optimize muted media away, which defeats the
+// purpose of holding the session open.
+
+function buildSilentWavUrl(seconds = 1, sampleRate = 8000): string {
+  const numSamples = seconds * sampleRate;
+  const bytes = 44 + numSamples * 2;
+  const buffer = new ArrayBuffer(bytes);
+  const view = new DataView(buffer);
+  const write = (offset: number, str: string) => {
+    for (let i = 0; i < str.length; i++) view.setUint8(offset + i, str.charCodeAt(i));
+  };
+  write(0, 'RIFF');
+  view.setUint32(4, bytes - 8, true);
+  write(8, 'WAVE');
+  write(12, 'fmt ');
+  view.setUint32(16, 16, true);
+  view.setUint16(20, 1, true);
+  view.setUint16(22, 1, true);
+  view.setUint32(24, sampleRate, true);
+  view.setUint32(28, sampleRate * 2, true);
+  view.setUint16(32, 2, true);
+  view.setUint16(34, 16, true);
+  write(36, 'data');
+  view.setUint32(40, numSamples * 2, true);
+  return URL.createObjectURL(new Blob([buffer], { type: 'audio/wav' }));
+}
+
+export function AudioKeepAlive() {
+  const audioRef = useRef<HTMLAudioElement | null>(null);
+
+  useEffect(() => {
+    const url = buildSilentWavUrl(1, 8000);
+    const el = new Audio(url);
+    el.loop = true;
+    el.volume = 1;
+    el.preload = 'auto';
+    audioRef.current = el;
+
+    const tryPlay = () => {
+      if (!audioRef.current) return;
+      if (!audioRef.current.paused) return;
+      audioRef.current.play().catch((err) => {
+        debug.log('[AudioKeepAlive] play blocked (will retry on next gesture):', err);
+      });
+    };
+
+    tryPlay();
+
+    // Autoplay may be blocked until first user interaction — re-attempt then.
+    const onGesture = () => tryPlay();
+    window.addEventListener('pointerdown', onGesture, { once: false });
+    window.addEventListener('keydown', onGesture, { once: false });
+
+    // If the webview ever pauses the element on background, resume on return.
+    const onWake = () => {
+      if (!document.hidden) tryPlay();
+    };
+    document.addEventListener('visibilitychange', onWake);
+    window.addEventListener('focus', onWake);
+    window.addEventListener('pageshow', onWake);
+
+    return () => {
+      window.removeEventListener('pointerdown', onGesture);
+      window.removeEventListener('keydown', onGesture);
+      document.removeEventListener('visibilitychange', onWake);
+      window.removeEventListener('focus', onWake);
+      window.removeEventListener('pageshow', onWake);
+      el.pause();
+      el.src = '';
+      URL.revokeObjectURL(url);
+      audioRef.current = null;
+    };
+  }, []);
+
+  return null;
+}

backend/backends/__init__.py

@@ -177,7 +177,7 @@ def _get_qwen_model_configs() -> list[ModelConfig]:
     backend_type = get_backend_type()
     if backend_type == "mlx":
         repo_1_7b = "mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16"
-        repo_0_6b = "mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16"  # 0.6B not available in MLX, falls back
+        repo_0_6b = "mlx-community/Qwen3-TTS-12Hz-0.6B-Base-bf16"
     else:
         repo_1_7b = "Qwen/Qwen3-TTS-12Hz-1.7B-Base"
         repo_0_6b = "Qwen/Qwen3-TTS-12Hz-0.6B-Base"

backend/backends/mlx_backend.py

@@ -45,11 +45,9 @@ class MLXTTSBackend:
         Returns:
             HuggingFace Hub model ID for MLX
         """
-        # MLX model mapping
         mlx_model_map = {
             "1.7B": "mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16",
-            # 0.6B not yet converted to MLX format
-            "0.6B": "mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16",  # Fallback to 1.7B
+            "0.6B": "mlx-community/Qwen3-TTS-12Hz-0.6B-Base-bf16",
         }
 
         if model_size not in mlx_model_map:
@@ -195,6 +193,8 @@ class MLXTTSBackend:
 
         logger.info("Generating audio for text: %s", text)
 
+        model_name = f"qwen-tts-{self._current_model_size}"
+
         def _generate_sync():
             """Run synchronous generation in thread pool."""
             # MLX generate() returns a generator yielding GenerationResult objects
@@ -220,36 +220,40 @@ class MLXTTSBackend:
                 logger.warning("Regenerating without voice prompt.")
                 ref_audio = None
 
-            # Check if model supports voice cloning via generate method
-            # MLX API may support ref_audio parameter directly
-            try:
-                # Try with voice cloning parameters if supported
-                if ref_audio:
-                    # Check if generate accepts ref_audio parameter
-                    import inspect
+            # Model is loaded → weights are on disk. Force offline so
+            # lazy tokenizer/config lookups inside mlx_audio don't hang
+            # when the user is disconnected (issue #462).
+            with force_offline_if_cached(True, model_name):
+                # Check if model supports voice cloning via generate method
+                # MLX API may support ref_audio parameter directly
+                try:
+                    # Try with voice cloning parameters if supported
+                    if ref_audio:
+                        # Check if generate accepts ref_audio parameter
+                        import inspect
 
-                    sig = inspect.signature(self.model.generate)
-                    if "ref_audio" in sig.parameters:
-                        # Generate with voice cloning
-                        for result in self.model.generate(text, ref_audio=ref_audio, ref_text=ref_text, lang_code=lang):
-                            audio_chunks.append(np.array(result.audio))
-                            sample_rate = result.sample_rate
+                        sig = inspect.signature(self.model.generate)
+                        if "ref_audio" in sig.parameters:
+                            # Generate with voice cloning
+                            for result in self.model.generate(text, ref_audio=ref_audio, ref_text=ref_text, lang_code=lang):
+                                audio_chunks.append(np.array(result.audio))
+                                sample_rate = result.sample_rate
+                        else:
+                            # Fallback: generate without voice cloning
+                            for result in self.model.generate(text, lang_code=lang):
+                                audio_chunks.append(np.array(result.audio))
+                                sample_rate = result.sample_rate
                     else:
-                        # Fallback: generate without voice cloning
+                        # No voice prompt, generate normally
                         for result in self.model.generate(text, lang_code=lang):
                             audio_chunks.append(np.array(result.audio))
                             sample_rate = result.sample_rate
-                else:
-                    # No voice prompt, generate normally
+                except Exception as e:
+                    # If voice cloning fails, try without it
+                    logger.warning("Voice cloning failed, generating without voice prompt: %s", e)
                     for result in self.model.generate(text, lang_code=lang):
                         audio_chunks.append(np.array(result.audio))
                         sample_rate = result.sample_rate
-            except Exception as e:
-                # If voice cloning fails, try without it
-                logger.warning("Voice cloning failed, generating without voice prompt: %s", e)
-                for result in self.model.generate(text, lang_code=lang):
-                    audio_chunks.append(np.array(result.audio))
-                    sample_rate = result.sample_rate
 
             # Concatenate all chunks
             if audio_chunks:
@@ -343,6 +347,8 @@ class MLXSTTBackend:
         """
         await self.load_model_async(model_size)
 
+        progress_model_name = f"whisper-{self.model_size}"
+
         def _transcribe_sync():
             """Run synchronous transcription in thread pool."""
             # MLX Whisper transcription using generate method
@@ -351,7 +357,11 @@ class MLXSTTBackend:
             if language:
                 decode_options["language"] = language
 
-            result = self.model.generate(str(audio_path), **decode_options)
+            # Model is loaded → weights are on disk. Force offline so
+            # lazy tokenizer/config lookups don't hang when the user is
+            # disconnected (issue #462).
+            with force_offline_if_cached(True, progress_model_name):
+                result = self.model.generate(str(audio_path), **decode_options)
 
             # Extract text from result
             if isinstance(result, str):

backend/backends/pytorch_backend.py

@@ -172,13 +172,19 @@ class PyTorchTTSBackend:
                     # This shouldn't happen in practice, but handle it
                     return {"prompt": cached_prompt}, True
 
+        model_name = f"qwen-tts-{self._current_model_size}"
+
         def _create_prompt_sync():
             """Run synchronous voice prompt creation in thread pool."""
-            return self.model.create_voice_clone_prompt(
-                ref_audio=str(audio_path),
-                ref_text=reference_text,
-                x_vector_only_mode=False,
-            )
+            # Model is loaded → weights are on disk. Force offline so
+            # lazy tokenizer/config lookups inside qwen_tts don't hang
+            # when the user is disconnected (issue #462).
+            with force_offline_if_cached(True, model_name):
+                return self.model.create_voice_clone_prompt(
+                    ref_audio=str(audio_path),
+                    ref_text=reference_text,
+                    x_vector_only_mode=False,
+                )
 
         # Run blocking operation in thread pool
         voice_prompt_items = await asyncio.to_thread(_create_prompt_sync)
@@ -221,19 +227,24 @@ class PyTorchTTSBackend:
         # Load model
         await self.load_model_async(None)
 
+        model_name = f"qwen-tts-{self._current_model_size}"
+
         def _generate_sync():
             """Run synchronous generation in thread pool."""
             # Set seed if provided
             if seed is not None:
                 manual_seed(seed, self.device)
 
-            # Generate audio - this is the blocking operation
-            wavs, sample_rate = self.model.generate_voice_clone(
-                text=text,
-                voice_clone_prompt=voice_prompt,
-                language=LANGUAGE_CODE_TO_NAME.get(language, "auto"),
-                instruct=instruct,
-            )
+            # Model is loaded → weights are on disk. Force offline so
+            # lazy tokenizer/config lookups inside qwen_tts don't hang
+            # when the user is disconnected (issue #462).
+            with force_offline_if_cached(True, model_name):
+                wavs, sample_rate = self.model.generate_voice_clone(
+                    text=text,
+                    voice_clone_prompt=voice_prompt,
+                    language=LANGUAGE_CODE_TO_NAME.get(language, "auto"),
+                    instruct=instruct,
+                )
             return wavs[0], sample_rate
 
         # Run blocking inference in thread pool to avoid blocking event loop
@@ -331,40 +342,46 @@ class PyTorchSTTBackend:
         """
         await self.load_model_async(model_size)
 
+        progress_model_name = f"whisper-{self.model_size}"
+
         def _transcribe_sync():
             """Run synchronous transcription in thread pool."""
             # Load audio
-            audio, sr = load_audio(audio_path, sample_rate=16000)
+            audio, _sr = load_audio(audio_path, sample_rate=16000)
 
-            # Process audio
-            inputs = self.processor(
-                audio,
-                sampling_rate=16000,
-                return_tensors="pt",
-            )
-            inputs = inputs.to(self.device)
-
-            # Generate transcription
-            # If language is provided, force it; otherwise let Whisper auto-detect
-            generate_kwargs = {}
-            if language:
-                forced_decoder_ids = self.processor.get_decoder_prompt_ids(
-                    language=language,
-                    task="transcribe",
+            # Model is loaded → weights are on disk. Force offline so
+            # `get_decoder_prompt_ids` and any lazy tokenizer lookups
+            # don't hang when the user is disconnected (issue #462).
+            with force_offline_if_cached(True, progress_model_name):
+                # Process audio
+                inputs = self.processor(
+                    audio,
+                    sampling_rate=16000,
+                    return_tensors="pt",
                 )
-                generate_kwargs["forced_decoder_ids"] = forced_decoder_ids
+                inputs = inputs.to(self.device)
 
-            with torch.no_grad():
-                predicted_ids = self.model.generate(
-                    inputs["input_features"],
-                    **generate_kwargs,
-                )
+                # Generate transcription
+                # If language is provided, force it; otherwise let Whisper auto-detect
+                generate_kwargs = {}
+                if language:
+                    forced_decoder_ids = self.processor.get_decoder_prompt_ids(
+                        language=language,
+                        task="transcribe",
+                    )
+                    generate_kwargs["forced_decoder_ids"] = forced_decoder_ids
 
-            # Decode
-            transcription = self.processor.batch_decode(
-                predicted_ids,
-                skip_special_tokens=True,
-            )[0]
+                with torch.no_grad():
+                    predicted_ids = self.model.generate(
+                        inputs["input_features"],
+                        **generate_kwargs,
+                    )
+
+                # Decode
+                transcription = self.processor.batch_decode(
+                    predicted_ids,
+                    skip_special_tokens=True,
+                )[0]
 
             return transcription.strip()
 

backend/backends/qwen_custom_voice_backend.py

@@ -28,6 +28,7 @@ from .base import (
     combine_voice_prompts as _combine_voice_prompts,
     model_load_progress,
 )
+from ..utils.hf_offline_patch import force_offline_if_cached
 
 logger = logging.getLogger(__name__)
 
@@ -104,18 +105,19 @@ class QwenCustomVoiceBackend:
             model_path = self._get_model_path(model_size)
             logger.info("Loading Qwen CustomVoice %s on %s...", model_size, self.device)
 
-            if self.device == "cpu":
-                self.model = Qwen3TTSModel.from_pretrained(
-                    model_path,
-                    torch_dtype=torch.float32,
-                    low_cpu_mem_usage=False,
-                )
-            else:
-                self.model = Qwen3TTSModel.from_pretrained(
-                    model_path,
-                    device_map=self.device,
-                    torch_dtype=torch.bfloat16,
-                )
+            with force_offline_if_cached(is_cached, model_name):
+                if self.device == "cpu":
+                    self.model = Qwen3TTSModel.from_pretrained(
+                        model_path,
+                        torch_dtype=torch.float32,
+                        low_cpu_mem_usage=False,
+                    )
+                else:
+                    self.model = Qwen3TTSModel.from_pretrained(
+                        model_path,
+                        device_map=self.device,
+                        torch_dtype=torch.bfloat16,
+                    )
 
         self._current_model_size = model_size
         self.model_size = model_size
@@ -184,6 +186,7 @@ class QwenCustomVoiceBackend:
         await self.load_model_async(None)
 
         speaker = voice_prompt.get("preset_voice_id") or QWEN_CV_DEFAULT_SPEAKER
+        model_name = f"qwen-custom-voice-{self._current_model_size}"
 
         def _generate_sync():
             if seed is not None:
@@ -203,7 +206,11 @@ class QwenCustomVoiceBackend:
             if instruct:
                 kwargs["instruct"] = instruct
 
-            wavs, sample_rate = self.model.generate_custom_voice(**kwargs)
+            # Model is loaded → weights are on disk. Force offline so
+            # lazy tokenizer/config lookups inside qwen_tts don't hang
+            # when the user is disconnected (issue #462).
+            with force_offline_if_cached(True, model_name):
+                wavs, sample_rate = self.model.generate_custom_voice(**kwargs)
             return wavs[0], sample_rate
 
         audio, sample_rate = await asyncio.to_thread(_generate_sync)

backend/requirements-mlx.txt

@@ -3,6 +3,12 @@
 
 mlx>=0.30.0
 
+# miniaudio is a runtime dep of mlx-audio's STT path (mlx_audio.stt).
+# mlx-audio itself is installed --no-deps (see comment below), so we
+# must list miniaudio explicitly here or transcription fails on fresh
+# M1 installs with `ModuleNotFoundError: miniaudio` (issue #505).
+miniaudio>=1.59
+
 # NOTE: mlx-audio is intentionally not listed here. From 0.3.1 onward it
 # declares `transformers==5.0.0rc3` / `>=5.0.0`, which conflicts with the
 # `transformers<=4.57.6` cap in requirements.txt and breaks CI's clean
@@ -10,6 +16,7 @@ mlx>=0.30.0
 # mlx_audio.stt.load) works fine on transformers 4.57.x in practice.
 #
 # Install it via `pip install --no-deps mlx-audio==0.4.1` after this file
-# (see .github/workflows/release.yml). All other mlx-audio runtime deps
-# (huggingface_hub, librosa, miniaudio, mlx-lm, numba, numpy, protobuf,
-# pyloudnorm, sounddevice, tqdm) are already in requirements.txt.
+# (see .github/workflows/release.yml). Most other mlx-audio runtime deps
+# (huggingface_hub, librosa, mlx-lm, numba, numpy, protobuf, pyloudnorm,
+# sounddevice, tqdm) are already in requirements.txt or pulled in by
+# other engines.

backend/tests/test_audio_preprocess.py

@@ -0,0 +1,112 @@
+"""
+Unit tests for reference-audio preprocessing.
+
+Covers :func:`backend.utils.audio.preprocess_reference_audio` and
+:func:`backend.utils.audio.validate_and_load_reference_audio`.
+"""
+
+import sys
+from pathlib import Path
+
+import numpy as np
+import pytest
+import soundfile as sf
+
+sys.path.insert(0, str(Path(__file__).parent.parent))
+
+from utils.audio import (  # noqa: E402
+    preprocess_reference_audio,
+    validate_and_load_reference_audio,
+)
+
+
+SR = 24000
+
+
+def _tone(duration_s: float, amp: float = 0.3, freq: float = 220.0) -> np.ndarray:
+    n = int(duration_s * SR)
+    t = np.arange(n, dtype=np.float32) / SR
+    return (amp * np.sin(2 * np.pi * freq * t)).astype(np.float32)
+
+
+def test_peak_cap_scales_hot_input():
+    audio = _tone(3.0, amp=0.99)
+    out = preprocess_reference_audio(audio, SR)
+    assert np.abs(out).max() <= 0.951
+
+
+def test_peak_cap_leaves_moderate_input_untouched():
+    audio = _tone(3.0, amp=0.5)
+    out = preprocess_reference_audio(audio, SR)
+    assert np.isclose(np.abs(out).max(), 0.5, atol=1e-3)
+
+
+def test_dc_offset_removed():
+    audio = _tone(3.0, amp=0.3) + 0.1
+    out = preprocess_reference_audio(audio, SR)
+    assert abs(float(np.mean(out))) < 1e-3
+
+
+def test_silence_is_trimmed_with_padding_kept():
+    silence = np.zeros(int(SR * 1.0), dtype=np.float32)
+    speech = _tone(3.0, amp=0.3)
+    audio = np.concatenate([silence, speech, silence])
+    out = preprocess_reference_audio(audio, SR)
+    # Most of the 2s of leading/trailing silence should be gone, but the
+    # 3s of speech plus ~200ms of padding should remain.
+    assert len(audio) - len(out) >= SR, "expected >=1s of silence trimmed"
+    assert len(out) >= int(3.0 * SR), "speech body should be preserved"
+
+
+def test_clean_audio_is_not_padded_past_original_length():
+    # Well-recorded audio with no edge silence shouldn't get longer after
+    # preprocessing — otherwise a 29.9 s upload could be pushed past the
+    # 30 s max_duration ceiling downstream.
+    audio = _tone(3.0, amp=0.3)
+    out = preprocess_reference_audio(audio, SR)
+    assert len(out) <= len(audio)
+
+
+def test_empty_input_returns_empty():
+    out = preprocess_reference_audio(np.zeros(0, dtype=np.float32), SR)
+    assert out.size == 0
+
+
+def test_validate_accepts_previously_rejected_hot_file(tmp_path):
+    audio = _tone(3.0, amp=0.995)
+    path = tmp_path / "hot.wav"
+    sf.write(str(path), audio, SR)
+
+    ok, err, out_audio, out_sr = validate_and_load_reference_audio(str(path))
+
+    assert ok, f"expected pass, got error: {err}"
+    assert out_audio is not None
+    assert out_sr == SR
+    assert np.abs(out_audio).max() <= 0.951
+
+
+def test_validate_still_rejects_silent_input(tmp_path):
+    audio = np.zeros(int(SR * 3.0), dtype=np.float32)
+    path = tmp_path / "silent.wav"
+    sf.write(str(path), audio, SR)
+
+    ok, err, _, _ = validate_and_load_reference_audio(str(path))
+
+    assert not ok
+    assert err is not None
+    assert "too short" in err.lower() or "quiet" in err.lower()
+
+
+def test_validate_rejects_too_short(tmp_path):
+    audio = _tone(0.5, amp=0.3)
+    path = tmp_path / "short.wav"
+    sf.write(str(path), audio, SR)
+
+    ok, err, _, _ = validate_and_load_reference_audio(str(path))
+
+    assert not ok
+    assert "too short" in (err or "").lower()
+
+
+if __name__ == "__main__":
+    pytest.main([__file__, "-v"])

backend/tests/test_offline_guard.py

@@ -0,0 +1,118 @@
+"""
+Unit tests for the ``force_offline_if_cached`` helper.
+
+Verifies that the helper mutates the cached module constants in
+``huggingface_hub.constants`` and ``transformers.utils.hub`` — not just
+``os.environ`` — and that concurrent users are refcount-coordinated so
+one thread's exit can't strip another thread's offline protection.
+
+NOTE: These tests mutate process-global state in ``huggingface_hub.constants``
+and ``transformers.utils.hub``. They are not safe under cross-process
+parallelism (e.g. ``pytest-xdist`` with ``--dist=loadfile``/``loadscope``);
+run this file serially.
+"""
+
+import os
+import sys
+import threading
+from pathlib import Path
+
+import pytest
+
+sys.path.insert(0, str(Path(__file__).parent.parent))
+
+from utils.hf_offline_patch import force_offline_if_cached  # noqa: E402
+
+
+def _hf_const():
+    import huggingface_hub.constants as hf_const
+
+    return hf_const
+
+
+def _tf_hub():
+    import transformers.utils.hub as tf_hub
+
+    return tf_hub
+
+
+def test_mutates_cached_huggingface_hub_constant():
+    original = _hf_const().HF_HUB_OFFLINE
+    with force_offline_if_cached(True, "t"):
+        assert _hf_const().HF_HUB_OFFLINE is True
+    assert original == _hf_const().HF_HUB_OFFLINE
+
+
+def test_mutates_cached_transformers_constant():
+    original = _tf_hub()._is_offline_mode
+    with force_offline_if_cached(True, "t"):
+        assert _tf_hub()._is_offline_mode is True
+    assert original == _tf_hub()._is_offline_mode
+
+
+def test_sets_env_variable():
+    original = os.environ.get("HF_HUB_OFFLINE")
+    with force_offline_if_cached(True, "t"):
+        assert "1" == os.environ.get("HF_HUB_OFFLINE")
+    assert original == os.environ.get("HF_HUB_OFFLINE")
+
+
+def test_noop_when_not_cached():
+    before = _hf_const().HF_HUB_OFFLINE
+    with force_offline_if_cached(False, "t"):
+        assert before == _hf_const().HF_HUB_OFFLINE
+
+
+def test_nested_contexts_respect_refcount():
+    original = _hf_const().HF_HUB_OFFLINE
+    with force_offline_if_cached(True, "outer"):
+        assert _hf_const().HF_HUB_OFFLINE is True
+        with force_offline_if_cached(True, "inner"):
+            assert _hf_const().HF_HUB_OFFLINE is True
+        # inner exit must not restore while outer is still active
+        assert _hf_const().HF_HUB_OFFLINE is True
+    assert original == _hf_const().HF_HUB_OFFLINE
+
+
+def test_concurrent_threads_share_offline_window():
+    """A slow thread must keep seeing offline mode even if a peer exits first."""
+    original = _hf_const().HF_HUB_OFFLINE
+    observations: list[bool] = []
+    errors: list[Exception] = []
+    barrier = threading.Barrier(2)
+    fast_exited = threading.Event()
+
+    def slow():
+        try:
+            with force_offline_if_cached(True, "slow"):
+                barrier.wait(timeout=5)
+                assert fast_exited.wait(timeout=5), "fast thread did not exit"
+                observations.append(_hf_const().HF_HUB_OFFLINE)
+        except Exception as exc:  # noqa: BLE001
+            errors.append(exc)
+
+    def fast():
+        try:
+            with force_offline_if_cached(True, "fast"):
+                barrier.wait(timeout=5)
+        except Exception as exc:  # noqa: BLE001
+            errors.append(exc)
+        finally:
+            fast_exited.set()
+
+    t_slow = threading.Thread(target=slow)
+    t_fast = threading.Thread(target=fast)
+    t_slow.start()
+    t_fast.start()
+    t_slow.join(timeout=5)
+    t_fast.join(timeout=5)
+
+    assert not t_slow.is_alive(), "slow thread did not finish"
+    assert not t_fast.is_alive(), "fast thread did not finish"
+    assert not errors, errors
+    assert observations == [True], "slow thread lost offline protection"
+    assert original == _hf_const().HF_HUB_OFFLINE
+
+
+if __name__ == "__main__":
+    pytest.main([__file__, "-v"])

backend/tests/test_qwen_download.py

@@ -175,7 +175,7 @@ async def main():
     print("   ✅ Server is running")
 
     # Test model
-    model_name = "qwen-tts-0.6B"  # Note: 0.6B currently maps to 1.7B on MLX
+    model_name = "qwen-tts-0.6B"
     
     # Check current status
     print(f"\n📊 Checking status of {model_name}...")

backend/utils/audio.py

@@ -199,6 +199,66 @@ def trim_tts_output(
     return trimmed
 
 
+def preprocess_reference_audio(
+    audio: np.ndarray,
+    sample_rate: int,
+    peak_target: float = 0.95,
+    trim_top_db: float = 40.0,
+    edge_padding_ms: int = 100,
+) -> np.ndarray:
+    """
+    Clean up a reference-audio sample before validation/storage.
+
+    Removes DC offset, trims leading/trailing silence, and caps the peak so a
+    slightly-hot recording doesn't get rejected downstream as "clipping". The
+    goal is to accept reasonable real-world recordings — not to repair badly
+    distorted ones. True clipping artifacts inside the waveform can't be
+    recovered by peak scaling and will still sound bad.
+
+    Args:
+        audio: Mono audio array.
+        sample_rate: Sample rate of ``audio`` in Hz.
+        peak_target: Peak amplitude cap in [0, 1]. Applied only if the input
+            peak exceeds this value.
+        trim_top_db: Silence threshold for edge trimming, in dB below peak.
+            40 dB sits below normal speech dynamic range (≈30 dB) so soft
+            trailing syllables are preserved, while still catching obvious
+            leading/trailing silence. Lower values are more aggressive;
+            librosa's own default is 60.
+        edge_padding_ms: Milliseconds of padding to add back at each edge
+            *only if* trimming shortened the waveform, so TTS engines have a
+            brief silence to anchor on without ever making the output longer
+            than the input.
+
+    Returns:
+        Preprocessed audio array (float32).
+    """
+    audio = audio.astype(np.float32, copy=False)
+
+    if audio.size == 0:
+        return audio
+
+    audio = audio - float(np.mean(audio))
+
+    trimmed, _ = librosa.effects.trim(audio, top_db=trim_top_db)
+    if 0 < trimmed.size < audio.size:
+        pad_each = int(sample_rate * edge_padding_ms / 1000)
+        # Never pad past the original length — for near-max-duration uploads
+        # an unconditional pad would push them over the 30 s ceiling and
+        # trigger a spurious "too long" rejection.
+        headroom = (audio.size - trimmed.size) // 2
+        pad = min(pad_each, max(headroom, 0))
+        if pad > 0:
+            trimmed = np.pad(trimmed, (pad, pad), mode="constant")
+        audio = trimmed
+
+    peak = float(np.abs(audio).max())
+    if peak > peak_target and peak > 0:
+        audio = audio * (peak_target / peak)
+
+    return audio
+
+
 def validate_reference_audio(
     audio_path: str,
     min_duration: float = 2.0,
@@ -207,13 +267,13 @@ def validate_reference_audio(
 ) -> Tuple[bool, Optional[str]]:
     """
     Validate reference audio for voice cloning.
-    
+
     Args:
         audio_path: Path to audio file
         min_duration: Minimum duration in seconds
         max_duration: Maximum duration in seconds
         min_rms: Minimum RMS level
-        
+
     Returns:
         Tuple of (is_valid, error_message)
     """
@@ -231,26 +291,28 @@ def validate_and_load_reference_audio(
 ) -> Tuple[bool, Optional[str], Optional[np.ndarray], Optional[int]]:
     """
     Validate and load reference audio in a single pass.
-    
+
+    Applies :func:`preprocess_reference_audio` before checks so that
+    slightly-hot recordings aren't rejected as clipping. Duration and RMS
+    checks run on the preprocessed waveform.
+
     Returns:
         Tuple of (is_valid, error_message, audio_array, sample_rate)
     """
     try:
         audio, sr = load_audio(audio_path)
+        audio = preprocess_reference_audio(audio, sr)
         duration = len(audio) / sr
-        
+
         if duration < min_duration:
             return False, f"Audio too short (minimum {min_duration} seconds)", None, None
         if duration > max_duration:
             return False, f"Audio too long (maximum {max_duration} seconds)", None, None
-        
+
         rms = np.sqrt(np.mean(audio**2))
         if rms < min_rms:
             return False, "Audio is too quiet or silent", None, None
-        
-        if np.abs(audio).max() > 0.99:
-            return False, "Audio is clipping (reduce input gain)", None, None
-        
+
         return True, None, audio, sr
     except Exception as e:
         return False, f"Error validating audio: {str(e)}", None, None

backend/utils/hf_offline_patch.py

@@ -6,6 +6,7 @@ are already downloaded. Must be imported BEFORE mlx_audio.
 
 import logging
 import os
+import threading
 from contextlib import contextmanager
 from pathlib import Path
 from typing import Optional, Union
@@ -13,13 +14,33 @@ from typing import Optional, Union
 logger = logging.getLogger(__name__)
 
 
+# huggingface_hub reads ``HF_HUB_OFFLINE`` once at import time into
+# ``huggingface_hub.constants.HF_HUB_OFFLINE``; transformers mirrors that into
+# ``transformers.utils.hub._is_offline_mode`` at *its* import time. Toggling
+# ``os.environ`` after either module is imported does not flip those cached
+# bools, and the hot paths (``_http._default_backend_factory``,
+# ``transformers.utils.hub.is_offline_mode``) read the bools — not the env.
+# We mutate the cached constants directly, guarded by a refcount so
+# concurrent inference threads share a single offline window safely.
+
+_offline_lock = threading.RLock()
+_offline_refcount = 0
+_saved_env: Optional[str] = None
+_saved_hf_const: Optional[bool] = None
+_saved_transformers_const: Optional[bool] = None
+
+
 @contextmanager
 def force_offline_if_cached(is_cached: bool, model_label: str = ""):
-    """Context manager that sets ``HF_HUB_OFFLINE=1`` while loading a cached model.
+    """Force offline mode for the duration of a cached-model operation.
+
+    Flips ``HF_HUB_OFFLINE`` in the process env **and** in the cached bools
+    inside ``huggingface_hub.constants`` / ``transformers.utils.hub`` so HTTP
+    adapters and offline-mode checks actually see the change. Uses a refcount
+    so multiple concurrent inference threads share a single offline window
+    and the last one to exit restores state.
 
     If *is_cached* is ``False`` the block runs normally (network allowed).
-    If the offline load raises an error containing "offline" we automatically
-    retry with network access so a partially-cached model still works.
 
     Args:
         is_cached: Whether the model weights are already on disk.
@@ -29,34 +50,96 @@ def force_offline_if_cached(is_cached: bool, model_label: str = ""):
         yield
         return
 
-    original_value = os.environ.get("HF_HUB_OFFLINE")
-    os.environ["HF_HUB_OFFLINE"] = "1"
-    logger.info(
-        "[offline-guard] %s is cached — forcing HF_HUB_OFFLINE=1",
-        model_label or "model",
-    )
+    global _offline_refcount, _saved_env, _saved_hf_const, _saved_transformers_const
+
+    with _offline_lock:
+        if _offline_refcount == 0:
+            # Snapshot prior state, apply new state, roll back on *any*
+            # failure. Catching only ImportError here would let a partially
+            # broken install (RuntimeError, AttributeError from a half-init
+            # module, etc.) leave the cached HF constants mutated without
+            # bumping the refcount — a persistent offline leak that outlives
+            # the process and is miserable to debug.
+            prev_env = os.environ.get("HF_HUB_OFFLINE")
+            prev_hf: Optional[bool] = None
+            prev_tf: Optional[bool] = None
+            try:
+                try:
+                    import huggingface_hub.constants as hf_const
+
+                    prev_hf = hf_const.HF_HUB_OFFLINE
+                    hf_const.HF_HUB_OFFLINE = True
+                except ImportError:
+                    prev_hf = None
+
+                try:
+                    import transformers.utils.hub as tf_hub
+
+                    prev_tf = tf_hub._is_offline_mode
+                    tf_hub._is_offline_mode = True
+                except ImportError:
+                    prev_tf = None
+
+                os.environ["HF_HUB_OFFLINE"] = "1"
+            except BaseException:
+                # Roll back whatever we already changed, then re-raise so
+                # the caller sees the real failure.
+                if prev_hf is not None:
+                    try:
+                        import huggingface_hub.constants as hf_const
+
+                        hf_const.HF_HUB_OFFLINE = prev_hf
+                    except ImportError:
+                        pass
+                if prev_tf is not None:
+                    try:
+                        import transformers.utils.hub as tf_hub
+
+                        tf_hub._is_offline_mode = prev_tf
+                    except ImportError:
+                        pass
+                if prev_env is not None:
+                    os.environ["HF_HUB_OFFLINE"] = prev_env
+                else:
+                    os.environ.pop("HF_HUB_OFFLINE", None)
+                raise
+
+            _saved_env = prev_env
+            _saved_hf_const = prev_hf
+            _saved_transformers_const = prev_tf
+            logger.info(
+                "[offline-guard] %s is cached — forcing offline mode",
+                model_label or "model",
+            )
+        _offline_refcount += 1
 
     try:
         yield
-    except Exception as exc:
-        if "offline" in str(exc).lower():
-            logger.warning(
-                "[offline-guard] Offline load failed for %s, retrying with network: %s",
-                model_label or "model",
-                exc,
-            )
-            # Restore original env and retry — caller must wrap the load
-            # inside force_offline_if_cached so retrying here isn't possible.
-            # Instead, propagate a flag via the exception so the caller can
-            # decide.  For simplicity we just let it fall through to the
-            # finally block and re-raise.
-            raise
-        raise
     finally:
-        if original_value is not None:
-            os.environ["HF_HUB_OFFLINE"] = original_value
-        else:
-            os.environ.pop("HF_HUB_OFFLINE", None)
+        with _offline_lock:
+            _offline_refcount -= 1
+            if _offline_refcount == 0:
+                if _saved_env is not None:
+                    os.environ["HF_HUB_OFFLINE"] = _saved_env
+                else:
+                    os.environ.pop("HF_HUB_OFFLINE", None)
+                if _saved_hf_const is not None:
+                    try:
+                        import huggingface_hub.constants as hf_const
+
+                        hf_const.HF_HUB_OFFLINE = _saved_hf_const
+                    except ImportError:
+                        pass
+                if _saved_transformers_const is not None:
+                    try:
+                        import transformers.utils.hub as tf_hub
+
+                        tf_hub._is_offline_mode = _saved_transformers_const
+                    except ImportError:
+                        pass
+                _saved_env = None
+                _saved_hf_const = None
+                _saved_transformers_const = None
 
 
 def patch_huggingface_hub_offline():

docs/PROJECT_STATUS.md

@@ -309,11 +309,7 @@ Still reported. Users get stuck downloads, can't resume, offline mode edge cases
 
 **Fix path:** PR #443 addresses infinite offline retry. CustomVoice-specific download failures (#475, #445) need triage — likely related to frozen-binary import fixes in PR #438. TADA cluster (#336, #348) and macOS ARM import regressions (#287, #275, #304) need a dedicated triage pass.
 
-**Qwen 0.6B-downloads-1.7B reports:** **#485** (2026-04-19), **#423** (macOS M1), **#329**. Platform-dependent:
-
-- **On MLX (Apple Silicon) — not a bug.** `mlx-community` only publishes 1.7B-Base-bf16 weights, so the 0.6B Base option intentionally resolves to the same repo (`backend/backends/__init__.py:180` — `# 0.6B not available in MLX, falls back`). UX gap: the selector offers a size that doesn't exist on the active backend. Fix: (a) hide the 0.6B option on MLX, or (b) label it "0.6B (uses 1.7B on Apple Silicon)".
-- **On PyTorch (Windows/Linux/CUDA/ROCm/XPU/CPU) — real bug if reported.** Both 0.6B and 1.7B have distinct repos (`Qwen/Qwen3-TTS-12Hz-0.6B-Base` vs `-1.7B-Base`). Triage each report by platform before merging into the MLX cluster.
-- **Qwen CustomVoice (either platform)** — no fallback, both sizes always have dedicated repos.
+**Qwen 0.6B-downloads-1.7B reports:** **#485** (2026-04-19), **#423** (macOS M1), **#329**. Originally a stale-fallback bug: `mlx-community/Qwen3-TTS-12Hz-0.6B-Base-bf16` wasn't published when MLX support shipped, so the 0.6B slot was aliased to the 1.7B repo. The 0.6B bf16 conversion is live now and both `backend/backends/mlx_backend.py` and `backend/backends/__init__.py` point at their correct repos. Qwen CustomVoice is unaffected — it runs via PyTorch on all platforms, both sizes always have dedicated repos.
 
 ### Language Requests (ongoing)
 
@@ -393,7 +389,7 @@ Notable:
 | **#306** ("voice model"), **#389** ("New model"), **#473** ("New functionality") | Title-only issues, no content. Request details or close. |
 | **#309** | Uninstall/cleanup question. Answer and close. |
 | **#241** | "How to use in Colab" — support question, not a bug. |
-| **#423** / **#485** / **#329** | Platform-dependent. On MLX: not a bug (0.6B weights don't exist upstream, fallback is intentional — fix UX). On PyTorch: real bug if reproducible. Classify each by reporter's platform before deduping. |
+| **#423** / **#485** / **#329** | Stale MLX fallback to 1.7B repo — fixed; 0.6B bf16 conversion now live on `mlx-community`, registry points at correct repo on both backends. |
 | **#336** / **#348** | TADA download/registration cluster — triage together. |
 | **#287** / **#275** / **#304** | macOS ARM import regressions on new version — likely one root cause. |
 | **#292**, **#349** | Possibly already fixed by merged PRs (#321/#412 and #345). Verify + close. |

docs/content/docs/overview/troubleshooting.mdx

@@ -446,20 +446,6 @@ Restart the app to create a fresh database.
   huggingface-cli download Qwen/Qwen3-TTS-12Hz-1.7B-Base
   ```
 
-### Qwen 0.6B Downloads the Same Files as 1.7B on Apple Silicon
-
-**Symptoms:**
-- You select Qwen 0.6B on an Apple Silicon Mac and the download is the same size as 1.7B
-- Generation speed and VRAM usage match 1.7B, not the expected smaller model
-
-**Explanation:**
-This is intentional, not a bug. The MLX community only publishes `mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16` — there is no 0.6B MLX build. Voicebox's model registry falls back to the 1.7B weights when 0.6B is selected on MLX (see `backend/backends/__init__.py`).
-
-**Solution:**
-- On Apple Silicon, both size options use the 1.7B model — pick either.
-- If you specifically need a smaller model, switch to **Kokoro 82M** (~350 MB) or **LuxTTS** (~300 MB) — both CPU-realtime.
-- On Windows/Linux with PyTorch, 0.6B and 1.7B are distinct repos and behave differently.
-
 ### Wrong Model Version
 
 **Symptoms:**

landing/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@voicebox/landing",
-  "version": "0.4.1",
+  "version": "0.4.2",
   "description": "Landing page for voicebox.sh",
   "scripts": {
     "dev": "next dev --turbo",

landing/src/app/download/[platform]/route.ts

@@ -15,17 +15,32 @@ const PLATFORM_ALIAS: Record<string, string> = {
   windows: 'windows',
 };
 
+function getPublicOrigin(request: NextRequest): string {
+  const forwardedHost = request.headers.get('x-forwarded-host');
+  const forwardedProto = request.headers.get('x-forwarded-proto');
+
+  if (forwardedHost && forwardedProto) {
+    // Behind reverse proxies/CDNs, request.url can be an internal origin
+    // (for example localhost:8080). Prefer forwarded headers so redirects
+    // keep users on the public domain.
+    return `${forwardedProto}://${forwardedHost}`;
+  }
+
+  return new URL(request.url).origin;
+}
+
 export async function GET(
   request: NextRequest,
   { params }: { params: Promise<{ platform: string }> },
 ) {
+  const origin = getPublicOrigin(request);
   const { platform } = await params;
   // No prebuilt Linux binary yet — send straight to the build-from-source page.
   if (platform === 'linux') {
-    return NextResponse.redirect(new URL('/linux-install', request.url), 307);
+    return NextResponse.redirect(new URL('/linux-install', origin), 307);
   }
   const normalized = PLATFORM_ALIAS[platform];
-  const target = new URL('/download', request.url);
+  const target = new URL('/download', origin);
   if (normalized) target.searchParams.set('platform', normalized);
   return NextResponse.redirect(target, 307);
 }

package.json

@@ -1,6 +1,6 @@
 {
   "name": "voicebox",
-  "version": "0.4.1",
+  "version": "0.4.2",
   "private": true,
   "workspaces": [
     "app",

tauri/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@voicebox/tauri",
   "private": true,
-  "version": "0.4.1",
+  "version": "0.4.2",
   "type": "module",
   "scripts": {
     "dev": "vite",

tauri/src-tauri/Cargo.lock

@@ -5041,7 +5041,7 @@ checksum = "0b928f33d975fc6ad9f86c8f283853ad26bdd5b10b7f1542aa2fa15e2289105a"
 
 [[package]]
 name = "voicebox"
-version = "0.4.1"
+version = "0.4.2"
 dependencies = [
  "base64 0.22.1",
  "core-foundation-sys",

tauri/src-tauri/Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "voicebox"
-version = "0.4.1"
+version = "0.4.2"
 description = "A production-quality desktop app for Qwen3-TTS voice cloning and generation"
 authors = ["you"]
 license = ""

tauri/src-tauri/tauri.conf.json

@@ -1,7 +1,7 @@
 {
   "$schema": "https://schema.tauri.app/config/2",
   "productName": "Voicebox",
-  "version": "0.4.1",
+  "version": "0.4.2",
   "identifier": "sh.voicebox.app",
   "build": {
     "beforeDevCommand": "bun run dev",

web/package.json

@@ -1,7 +1,7 @@
 {
   "name": "@voicebox/web",
   "private": true,
-  "version": "0.4.1",
+  "version": "0.4.2",
   "type": "module",
   "scripts": {
     "dev": "vite",