fix(landing): address PR #487 review feedback

- Preserve canonical camelCase platform aliases (macArm, macIntel) in the
  /download/[platform] redirect so those URLs don't lose their platform param.
- Add accessible title + role="img" to the inline Windows SVG so it passes
  Biome's a11y rule and announces to screen readers.
- On /api/releases fetch failure, show an explicit error state with a single
  intentional link to GitHub releases — no more silent GitHub fallback or
  disabled-button UX lie. Keeps normies off GitHub unless they opt in.

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

app/src/components/AppFrame/AppFrame.tsx

@@ -1,6 +1,5 @@
 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';
@@ -27,7 +26,6 @@ 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

@@ -1,85 +0,0 @@
-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/tests/test_audio_preprocess.py

@@ -1,112 +0,0 @@
-"""
-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/utils/audio.py

@@ -199,66 +199,6 @@ 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,
@@ -267,13 +207,13 @@ 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)
     """
@@ -291,28 +231,26 @@ 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

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

@@ -15,32 +15,17 @@ 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', origin), 307);
+    return NextResponse.redirect(new URL('/linux-install', request.url), 307);
   }
   const normalized = PLATFORM_ALIAS[platform];
-  const target = new URL('/download', origin);
+  const target = new URL('/download', request.url);
   if (normalized) target.searchParams.set('platform', normalized);
   return NextResponse.redirect(target, 307);
 }