diff --git a/api/test/peerconnection_quality_test_fixture.h b/api/test/peerconnection_quality_test_fixture.h
index b5d3f3ce20..b60c3b4285 100644
--- a/api/test/peerconnection_quality_test_fixture.h
+++ b/api/test/peerconnection_quality_test_fixture.h
@@ -381,6 +381,10 @@ class PeerConnectionE2EQualityTestFixture {
     // According to bugs.webrtc.org/4762 WebRTC supports synchronization only
     // for pair of single audio and single video stream.
     absl::optional<std::string> sync_group;
+    // If specified, it will be set into RtpParameters of corresponding
+    // RtpSenderInterface for this video stream.
+    // Note that this setting takes precedence over `content_hint`.
+    absl::optional<DegradationPreference> degradation_preference;
   };
 
   // Contains properties for audio in the call.
diff --git a/test/pc/e2e/media/media_helper.cc b/test/pc/e2e/media/media_helper.cc
index 22fcc51ffe..4885a28ad0 100644
--- a/test/pc/e2e/media/media_helper.cc
+++ b/test/pc/e2e/media/media_helper.cc
@@ -83,11 +83,18 @@ MediaHelper::MaybeAddVideo(TestPeer* peer) {
     RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> sender =
         peer->AddTrack(track, {sync_group, *video_config.stream_label});
     RTC_CHECK(sender.ok());
-    if (video_config.temporal_layers_count) {
+    if (video_config.temporal_layers_count ||
+        video_config.degradation_preference) {
       RtpParameters rtp_parameters = sender.value()->GetParameters();
-      for (auto& encoding_parameters : rtp_parameters.encodings) {
-        encoding_parameters.num_temporal_layers =
-            video_config.temporal_layers_count;
+      if (video_config.temporal_layers_count) {
+        for (auto& encoding_parameters : rtp_parameters.encodings) {
+          encoding_parameters.num_temporal_layers =
+              video_config.temporal_layers_count;
+        }
+      }
+      if (video_config.degradation_preference) {
+        rtp_parameters.degradation_preference =
+            video_config.degradation_preference;
       }
       RTCError res = sender.value()->SetParameters(rtp_parameters);
       RTC_CHECK(res.ok()) << "Failed to set RTP parameters";