diff --git a/src/pages/Player/RealtimeWaveform.tsx b/src/pages/Player/RealtimeWaveform.tsx
index 4f302fc..342f4ec 100644
--- a/src/pages/Player/RealtimeWaveform.tsx
+++ b/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.
diff --git a/src/utils/audio-stream-reader.ts b/src/utils/audio-stream-reader.ts
index b6da4e8..0b3dc9d 100644
--- a/src/utils/audio-stream-reader.ts
+++ b/src/utils/audio-stream-reader.ts
@@ -7,10 +7,12 @@
  * and serves windows *at a requested playback position* to the caller.
  *
  * This is independent from the actual playback backend — it's a
- * read-only "tap" on the audio for FFT analysis purposes. Because it is a
- * separate decoder, sync with the player is maintained by pacing decode at
- * the player's clock rate (`-readrate <speed>`) and sampling the window at
- * the position the player reports, never at the decode head.
+ * read-only "tap" on the audio for FFT analysis purposes. Sync with the
+ * player is maintained by pacing decode at the player's clock rate
+ * (`-readrate <speed>`) while front-loading a burst of LEAD_SECONDS
+ * (`-readrate_initial_burst`) so the decode head leads the player
+ * position by a stable lead — read() samples at the exact position the
+ * player reports, never at the decode head.
  */
 
 /** PCM output format constants */
@@ -27,6 +29,26 @@ const BYTES_PER_SAMPLE = 2; // s16le
  */
 const RING_BUFFER_SAMPLES = SAMPLE_RATE * 10;
 
+/**
+ * Decode-head lead over the player position, in seconds.
+ *
+ * `-readrate_initial_burst LEAD_SECONDS` makes ffmpeg emit this much audio
+ * immediately on start, then pace at realtime (`-readrate speed`) after.
+ * The decode head thus leads the player by ~LEAD_SECONDS from the very
+ * first frame. read() samples at the player's current position, which is
+ * always behind the head — so it finds freshly decoded samples there
+ * instead of clamping to stale data.
+ *
+ * Bare `-readrate speed` (no burst) starts ffmpeg ε behind mpv (input-open
+ * + first-packet latency) and, since both advance at the same rate, never
+ * catches up — the bars lag by ε (up to several seconds on network
+ * streams). The burst eliminates that constant offset.
+ *
+ * Must stay within the ring window (RING_BUFFER_SAMPLES ~10s) so the
+ * lead audio hasn't wrapped out by the time the player reaches it.
+ */
+const LEAD_SECONDS = 3;
+
 export interface AudioStreamReaderOptions {
 	/** Audio URL or file path to decode */
 	url: string;
@@ -79,8 +101,10 @@ export class AudioStreamReader {
 	 * immediately.
 	 *
 	 * @param startPosition Seek position in seconds (default: 0).
-	 * @param speed Playback speed multiplier (default: 1). Applies ffmpeg
-	 *              atempo filter so visualization stays in sync with audio.
+	 * @param speed Playback speed multiplier (default: 1). Paces ffmpeg
+	 *              at the player's advance rate so decode tracks the
+	 *              player clock; `-readrate_initial_burst` front-loads
+	 *              a LEAD_SECONDS head start.
 	 */
 	start(startPosition = 0, speed = 1): void {
 		// Always kill the previous process first — no early return on _running
@@ -101,17 +125,18 @@ export class AudioStreamReader {
 			"ffmpeg",
 			"-loglevel",
 			"quiet",
-			// Pace input at the player's advance rate (speed× native) rather
-			// than native rate. Decoding slower than the player makes the
-			// decoded position fall behind the playback position linearly
-			// (bars drift away at (speed-1)s per second); decoding unthrottled
-			// fills the ring with audio seconds ahead of the player (laggy
-			// bars) and hits EOF early (bars freeze). `-readrate speed` keeps
-			// the decode head just ahead of the position the renderer samples,
-			// tracking the player clock with only mpv's startup latency as a
-			// constant offset — absorbed by the ring buffer.
+			// Pace input at the player's advance rate (speed× native). Combined
+			// with -readrate_initial_burst below, the decode head starts
+			// LEAD_SECONDS ahead of the player and advances at the same rate —
+			// read() samples at the player position and always finds fresh data.
 			"-readrate",
 			String(readRate),
+			// Front-load LEAD_SECONDS of audio immediately so the decode head
+			// leads the player from the very first frame. Without this, ffmpeg
+			// starts ε behind mpv (input-open + first-packet latency) and,
+			// pacing at the same rate, never catches up — bars lag by ε.
+			"-readrate_initial_burst",
+			String(LEAD_SECONDS),
 		];
 
 		// `-reconnect*` are http-protocol options: ffmpeg rejects them at
diff --git a/tests/audio-stream-reader.test.ts b/tests/audio-stream-reader.test.ts
index 456699d..d9974d9 100644
--- a/tests/audio-stream-reader.test.ts
+++ b/tests/audio-stream-reader.test.ts
@@ -170,3 +170,70 @@ test.skipIf(!hasFfmpeg)(
 		}
 	},
 );
+
+test.skipIf(!hasFfmpeg)(
+	"sustained render loop: ffmpeg stays alive and decode head maintains a lead over the player",
+	async () => {
+		// Real wall-clock time is required here: this test validates ffmpeg's
+		// actual decode pacing (-readrate + -readrate_initial_burst) against
+		// the platform clock. Deterministic time control cannot reproduce the
+		// race where ffmpeg exits early and the bars freeze — that only
+		// surfaces when a real process writes to a real pipe.
+		//
+		// Simulates the actual render loop: for ~5s of wall time, advance a
+		// simulated player position at 1× realtime and call read() each frame.
+		// The decode head must stay ahead of the player position so read()
+		// always returns 512 samples, and ffmpeg must not exit early (which
+		// would freeze the bars). This test would have caught the
+		// backpressure-pacing failure where ffmpeg decoded all data into the
+		// pipe buffer instantly, exited, and the readLoop stopped.
+		const wav = join(
+			tmpdir(),
+			`podtui-reader-${process.pid}-${Date.now()}.wav`,
+		);
+		writeSineWav(wav, 30);
+		const reader = new AudioStreamReader({ url: wav });
+		try {
+			reader.start(0, 1);
+
+			const FRAME_MS = 33;
+			const DURATION_MS = 5000;
+			const out = new Float64Array(512);
+			let successes = 0;
+			let failures = 0;
+			let minLead = Infinity;
+
+			const start = Date.now();
+			for (let frame = 0; Date.now() - start < DURATION_MS; frame++) {
+				const playerPos = (Date.now() - start) / 1000;
+				const count = reader.read(out, playerPos);
+				if (count === 512) successes++;
+				else failures++;
+
+				// The decode head should stay ahead of the player position.
+				const headPos = reader.samplesWritten / SAMPLE_RATE;
+				const lead = headPos - playerPos;
+				if (frame > 3) minLead = Math.min(minLead, lead);
+
+				await Bun.sleep(FRAME_MS);
+			}
+
+			// ffmpeg must still be running — it must not have exited early.
+			expect(reader.running).toBe(true);
+
+			// The vast majority of frames should return a full window.
+			// A few early failures during ffmpeg startup are acceptable.
+			expect(failures).toBeLessThan(5);
+			expect(successes).toBeGreaterThan(100);
+
+			// The decode head must maintain a positive lead over the player.
+			// Without -readrate_initial_burst, the head would lag behind by
+			// the ffmpeg startup latency and never catch up.
+			expect(minLead).toBeGreaterThan(0);
+		} finally {
+			reader.stop();
+			await Bun.$`rm -f ${wav}`.quiet();
+		}
+	},
+	{ timeout: 15000 },
+);