fix(waveform): add -readrate_initial_burst to eliminate audio lag

The decode head lagged the player by a constant ε (ffmpeg startup
latency) because ffmpeg paced at -readrate <speed> started behind mpv
and, advancing at the same rate, never caught up — bars were a few
seconds behind for the entire playback.

Add -readrate_initial_burst LEAD_SECONDS so ffmpeg emits 3s of audio
immediately on start, then paces at realtime after. The decode head
leads the player by a stable ~3s from the very first frame; read()
samples at the exact player position and always finds fresh samples.

Add a sustained render-loop test that simulates ~5s of real playback,
asserting ffmpeg stays alive, the decode head maintains a positive
lead, and read() returns full windows. Uses real wall-clock time
(documented exception) since ffmpeg's decode pacing can't be tested
deterministically.

src/pages/Player/RealtimeWaveform.tsx

@@ -168,8 +168,9 @@ export function RealtimeWaveform(props: RealtimeWaveformProps) {
 		if (!cava?.isReady || !reader?.running || !sampleBuffer) return;
 
 		// Sample the FFT window at the player's position, not the decode
-		// head — the reader decodes independently and only the position clock
-		// ties the bars to what's actually playing.
+		// head — the reader decodes independently (paced at the player's
+		// clock rate with a LEAD_SECONDS burst head start) and only the
+		// position clock ties the bars to what's actually playing.
 		const target = smoothPosition();
 		const count = reader.read(sampleBuffer, target);
 		// Never feed a partial FFT window to cava.