diff --git a/video/rtp_streams_synchronizer.cc b/video/rtp_streams_synchronizer.cc
index 28e9a0ba9d..29ace90431 100644
--- a/video/rtp_streams_synchronizer.cc
+++ b/video/rtp_streams_synchronizer.cc
@@ -89,11 +89,18 @@ void RtpStreamsSynchronizer::Process() {
     log_stats = true;
   }
 
+  int64_t last_audio_receive_time_ms =
+      audio_measurement_.latest_receive_time_ms;
   absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
   if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
     return;
   }
 
+  if (last_audio_receive_time_ms == audio_measurement_.latest_receive_time_ms) {
+    // No new audio packet has been received since last update.
+    return;
+  }
+
   int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
   absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
   if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
diff --git a/video/rtp_streams_synchronizer2.cc b/video/rtp_streams_synchronizer2.cc
index cc084bc0aa..4096fceb99 100644
--- a/video/rtp_streams_synchronizer2.cc
+++ b/video/rtp_streams_synchronizer2.cc
@@ -92,11 +92,18 @@ void RtpStreamsSynchronizer::UpdateDelay() {
     log_stats = true;
   }
 
+  int64_t last_audio_receive_time_ms =
+      audio_measurement_.latest_receive_time_ms;
   absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
   if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
     return;
   }
 
+  if (last_audio_receive_time_ms == audio_measurement_.latest_receive_time_ms) {
+    // No new audio packet has been received since last update.
+    return;
+  }
+
   int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
   absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
   if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
diff --git a/video/stream_synchronization_unittest.cc b/video/stream_synchronization_unittest.cc
index 04a43c21f9..3d6fdd82a7 100644
--- a/video/stream_synchronization_unittest.cc
+++ b/video/stream_synchronization_unittest.cc
@@ -383,6 +383,63 @@ TEST_F(StreamSynchronizationTest, AudioDelayed) {
             total_audio_delay_ms);
 }
 
+TEST_F(StreamSynchronizationTest, NoAudioIncomingUnboundedIncrease) {
+  // Test how audio delay can grow unbounded when audio stops coming in.
+  // This is handled in caller of RtpStreamsSynchronizer, for example in
+  // RtpStreamsSynchronizer by not updating delays when audio samples stop
+  // coming in.
+  const int kVideoDelayMs = 300;
+  const int kAudioDelayMs = 100;
+  int current_audio_delay_ms = kAudioDelayMs;
+  int total_audio_delay_ms = 0;
+  int total_video_delay_ms = 0;
+
+  EXPECT_TRUE(DelayedStreams(/*audio_delay_ms=*/0, kVideoDelayMs,
+                             current_audio_delay_ms, &total_audio_delay_ms,
+                             &total_video_delay_ms));
+  EXPECT_EQ(0, total_video_delay_ms);
+  // The delay is not allowed to change more than this.
+  EXPECT_EQ((kVideoDelayMs - kAudioDelayMs) / kSmoothingFilter,
+            total_audio_delay_ms);
+  int last_total_audio_delay_ms = total_audio_delay_ms;
+
+  // Set new current audio delay: simulate audio samples are flowing in.
+  current_audio_delay_ms = total_audio_delay_ms;
+
+  clock_sender_.AdvanceTimeMilliseconds(1000);
+  clock_receiver_.AdvanceTimeMilliseconds(1000);
+  EXPECT_TRUE(DelayedStreams(/*audio_delay_ms=*/0, kVideoDelayMs,
+                             current_audio_delay_ms, &total_audio_delay_ms,
+                             &total_video_delay_ms));
+  EXPECT_EQ(0, total_video_delay_ms);
+  EXPECT_EQ(last_total_audio_delay_ms +
+                MaxAudioDelayChangeMs(current_audio_delay_ms, kVideoDelayMs),
+            total_audio_delay_ms);
+  last_total_audio_delay_ms = total_audio_delay_ms;
+
+  // Simulate no incoming audio by not update audio delay.
+  const int kSimulationSecs = 300;     // 5min
+  const int kMaxDeltaDelayMs = 10000;  // max delay for audio in webrtc
+  for (auto time_secs = 0; time_secs < kSimulationSecs; time_secs++) {
+    clock_sender_.AdvanceTimeMilliseconds(1000);
+    clock_receiver_.AdvanceTimeMilliseconds(1000);
+    EXPECT_TRUE(DelayedStreams(/*audio_delay_ms=*/0, kVideoDelayMs,
+                               current_audio_delay_ms, &total_audio_delay_ms,
+                               &total_video_delay_ms));
+    EXPECT_EQ(0, total_video_delay_ms);
+
+    // Audio delay does not go above kMaxDeltaDelayMs.
+    EXPECT_EQ(std::min(kMaxDeltaDelayMs,
+                       last_total_audio_delay_ms +
+                           MaxAudioDelayChangeMs(current_audio_delay_ms,
+                                                 kVideoDelayMs)),
+              total_audio_delay_ms);
+    last_total_audio_delay_ms = total_audio_delay_ms;
+  }
+  // By now the audio delay has grown unbounded to kMaxDeltaDelayMs.
+  EXPECT_EQ(kMaxDeltaDelayMs, last_total_audio_delay_ms);
+}
+
 TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) {
   BothDelayedVideoLaterTest(0);
 }