fix: always apply aresample sync filter and tighten audio sync tolerances

- Always route audio through aresample=async=1:first_pts=0 filter during
  muxing on both macOS and Windows, even when duration delta is small.
  Previously skipped when delta ≤50ms, which left progressive clock drift
  (from CPU load) completely uncorrected.
- Lower sync detection threshold from 50ms to 20ms so tempo correction
  kicks in earlier.
- Use explicit 48kHz AudioContext sample rate in both browser recording
  and export rendering to prevent sample rate mismatch drift.
- Reduce MediaRecorder timeslice from 1000ms to 250ms to reduce chunk
  loss under CPU pressure.
- Tighten export audio sync: seek threshold 300ms→150ms, tolerance
  15ms→8ms, correction window 2s→0.5s, max adjustment ±8%→±12%.
- Lower companion audio start delay threshold from 50ms to 25ms.

src/hooks/useScreenRecorder.ts

@@ -18,7 +18,7 @@ const HIGH_FRAME_RATE_BOOST = 1.7;
 const DEFAULT_WIDTH = 1920;
 const DEFAULT_HEIGHT = 1080;
 const CODEC_ALIGNMENT = 2;
-const RECORDER_TIMESLICE_MS = 1000;
+const RECORDER_TIMESLICE_MS = 250;
 const BITS_PER_MEGABIT = 1_000_000;
 const MIN_FRAME_RATE = 30;
 const CHROME_MEDIA_SOURCE = "desktop";
@@ -1090,7 +1090,7 @@ export function useScreenRecorder(): UseScreenRecorderReturn {
 				const micAudioTrack = microphoneStream.current?.getAudioTracks()[0];
 
 				if (systemAudioTrack && micAudioTrack) {
-					const context = new AudioContext();
+					const context = new AudioContext({ sampleRate: 48000 });
 					mixingContext.current = context;
 					const systemSource = context.createMediaStreamSource(
 						new MediaStream([systemAudioTrack]),