Fix audio buffer handling to reduce playback stalls and desync

README.md

@@ -79,7 +79,7 @@ A source you just added sizes itself to the canvas when its first frame arrives,
 |---|---|---|
 | URL | | Your pull URL. SRT, RTMP, or anything else FFmpeg can open |
 | Reconnect Delay | 2s | How long to wait between reconnect attempts |
-| Target Buffer | 120ms | How much audio cushion to hold. This is your main latency knob: higher rides out a worse connection, lower is snappier and less forgiving |
+| Target Buffer | 120ms | How much audio cushion to hold, 20ms to 2s. This is your main latency knob: higher rides out a worse connection, lower is snappier and less forgiving. If the stats show `underruns` climbing, this is the setting to raise — an underrun means the cushion ran dry, and the concealment that covers it delays video by the same amount to keep lip sync |
 | Adaptive Latency Control | On | Holds latency near your target by nudging playback speed (up to 2% slow, 5% fast) instead of dropping audio |
 | FFmpeg Options | | Extra options for the stream reader, `key1=val1 key2=val2` style. Use this to set the SRT `latency`, for example |
 | Hardware Decode | Auto | Let the GPU decode video. Auto picks whatever your machine supports, Off forces the CPU |

include/irl-source.h

@@ -151,7 +151,7 @@ struct irl_source;
  * are due, which is exactly the old behaviour, and counted so it is visible.
  */
 #define IRL_VIDEO_PACING_MAX_FRAMES 512
-#define IRL_VIDEO_PACING_MAX_BYTES (192u * 1024u * 1024u)
+#define IRL_VIDEO_PACING_MAX_BYTES (512u * 1024u * 1024u)
 /* Emit rather than sleep again when this close to due: another wakeup costs
  * more than the timing error it would remove. */
 #define IRL_VIDEO_PACING_SLACK_NS 1000000LL

src/receiver.c

@@ -83,12 +83,23 @@ void *irl_receiver_thread(void *data)
 		 * buffer to drop audible data. */
 		if (ctx->audio_stream_idx >= 0 &&
 		    !ctx->config.low_latency_audio) {
-			int pace_ms =
-				(int)os_atomic_load_long(
-					&ctx->config.buffer_max_ms) *
-				3;
+			int buffer_max_ms = (int)os_atomic_load_long(
+				&ctx->config.buffer_max_ms);
+			int pace_ms = buffer_max_ms * 3;
 			if (pace_ms > IRL_BLEED_PACE_FILL_MS)
 				pace_ms = IRL_BLEED_PACE_FILL_MS;
+			/* The flat cap above is an absolute latency guard, but
+			 * it must never fall to where playback cannot prime:
+			 * priming waits for target + the OBS output lead, and a
+			 * ceiling below that stops the read loop before the
+			 * buffer ever reaches it, so the source would sit
+			 * silent forever. buffer_max is target + 200, so this
+			 * floor clears the prime threshold by ~220ms at every
+			 * target. Only binds above a ~700ms target, which is
+			 * why nothing hit it while the setting stopped at
+			 * 500ms. */
+			if (pace_ms < buffer_max_ms + 100)
+				pace_ms = buffer_max_ms + 100;
 			while (os_atomic_load_bool(&ctx->thread_active) &&
 			       audio_buffer_fill_ms_locked(&ctx->audio_buf) >
 				       pace_ms) {

src/settings.c

@@ -60,8 +60,13 @@ obs_properties_t *irl_source_get_properties(void *data)
 
 	/* ── Audio Buffer ──────────────────────────────────── */
 
+	/* Up to 2s: IRL uplinks routinely stall for over a second (a field log
+	 * showed 1.7s gaps with 287 underruns at the 120ms default), and
+	 * riding those out is the only way to avoid the concealment that
+	 * inflates the A/V mapping and holds video back with it. The old 500ms
+	 * limit could not cover them. */
 	obs_properties_add_int(props, "buffer_target_ms",
-			       obs_module_text("Target Buffer (ms)"), 20, 500,
+			       obs_module_text("Target Buffer (ms)"), 20, 2000,
 			       10);
 	obs_properties_add_bool(props, "adaptive_speed",
 				obs_module_text("Adaptive Latency Control"));