fix: improve audio timestamp handling in buffered mode to reduce jitter

Author: datagutt

src/receiver.c

@@ -329,14 +329,25 @@ static uint64_t next_audio_timestamp(struct irl_source *ctx, int base_samples,
 			ctx->audio_sys_base = (uint64_t)(base > 0 ? base : 0);
 			audio_ts = now;
 		}
-	} else if (drift > 30000000LL) {
-		ctx->audio_pll_corrections++;
-		ctx->audio_pll_offset_ns -= frame_ns;
-		audio_ts -= frame_ns;
-	} else if (drift < -30000000LL) {
-		ctx->audio_pll_corrections++;
-		ctx->audio_sys_base += (uint64_t)frame_ns;
-		audio_ts += frame_ns;
+	} else {
+		/* Buffered mode should keep a small amount of audio queued
+		 * inside OBS, otherwise OBS sees a just-in-time stream and
+		 * grows its own async buffer on the first jitter spike. */
+		int64_t target_lead_ns = frame_ns * 2;
+		int64_t min_drift = target_lead_ns - 30000000LL;
+		int64_t max_drift = target_lead_ns + 30000000LL;
+
+		if (drift < min_drift) {
+			ctx->audio_pll_corrections++;
+			int64_t target_ts = (int64_t)now + target_lead_ns;
+			int64_t base = target_ts - ctx->audio_pll_offset_ns;
+			ctx->audio_sys_base = (uint64_t)(base > 0 ? base : 0);
+			audio_ts = (uint64_t)(target_ts > 0 ? target_ts : 0);
+		} else if (drift > max_drift) {
+			ctx->audio_pll_corrections++;
+			ctx->audio_pll_offset_ns -= frame_ns;
+			audio_ts -= frame_ns;
+		}
 	}
 
 	drift = (int64_t)audio_ts - (int64_t)now;
@@ -801,11 +812,9 @@ static void handle_audio_frame(struct irl_source *ctx, AVFrame *frame)
 		 * next_audio_ts_min because both advance by exactly
 		 * base_samples / sample_rate per push.
 		 *
-		 * Gentle PLL: when buffered mode drifts more than
-		 * ~30ms ahead or behind wall clock, slew by one
-		 * frame so OBS sees a continuous timeline instead
-		 * of a large jump. Low-latency mode still snaps
-		 * directly to wall clock. */
+		 * Buffered mode keeps roughly two chunks queued in OBS
+		 * so async audio doesn't immediately grow its own buffer.
+		 * Low-latency mode still snaps directly to wall clock. */
 		uint32_t frames_out =
 			(uint32_t)(got /
 				   (out_channels * bytes_per_sample));