diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 26805539b9..e5052bee7b 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -26,12 +26,14 @@ LayerFilteringTransport::LayerFilteringTransport(
     uint8_t vp8_video_payload_type,
     uint8_t vp9_video_payload_type,
     int selected_tl,
-    int selected_sl)
+    int selected_sl,
+    const std::set<uint32_t> &excluded_ssrcs)
     : test::DirectTransport(config, send_call),
       vp8_video_payload_type_(vp8_video_payload_type),
       vp9_video_payload_type_(vp9_video_payload_type),
       selected_tl_(selected_tl),
       selected_sl_(selected_sl),
+      excluded_ssrcs_(std::move(excluded_ssrcs)),
       discarded_last_packet_(false) {}
 
 bool LayerFilteringTransport::DiscardedLastPacket() const {
@@ -81,7 +83,9 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
       } else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
                   temporal_idx > selected_tl_) ||
                  (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
-                  spatial_idx > selected_sl_)) {
+                  spatial_idx > selected_sl_) ||
+                 excluded_ssrcs_.find(header.ssrc) !=
+                     excluded_ssrcs_.end()) {
         // Truncate packet to a padding packet.
         length = header.headerLength + 1;
         temp_buffer[0] |= (1 << 5);  // P = 1.
diff --git a/webrtc/test/layer_filtering_transport.h b/webrtc/test/layer_filtering_transport.h
index 8f8cc2e313..c2ab56fa00 100644
--- a/webrtc/test/layer_filtering_transport.h
+++ b/webrtc/test/layer_filtering_transport.h
@@ -27,7 +27,8 @@ class LayerFilteringTransport : public test::DirectTransport {
                           uint8_t vp8_video_payload_type,
                           uint8_t vp9_video_payload_type,
                           int selected_tl,
-                          int selected_sl);
+                          int selected_sl,
+                          const std::set<uint32_t> &excluded_ssrcs_);
   bool DiscardedLastPacket() const;
   bool SendRtp(const uint8_t* data,
                size_t length,
@@ -41,6 +42,7 @@ class LayerFilteringTransport : public test::DirectTransport {
   // selected one. -1 to disable filtering.
   const int selected_tl_;
   const int selected_sl_;
+  const std::set<uint32_t> excluded_ssrcs_;
   bool discarded_last_packet_;
 };
 
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 6818a5cad6..920076d865 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1438,21 +1438,26 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
   call_config.bitrate_config = params.call.call_bitrate_config;
   CreateCalls(call_config, call_config);
 
-  test::LayerFilteringTransport send_transport(
-      params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
-      params_.video.selected_tl, params_.ss.selected_sl);
-  test::DirectTransport recv_transport(params_.pipe, receiver_call_.get());
+  VideoStream& selected_stream = params_.ss.streams[params_.ss.selected_stream];
 
   std::string graph_title = params_.analyzer.graph_title;
   if (graph_title.empty())
     graph_title = VideoQualityTest::GenerateGraphTitle();
 
-  // In the case of different resolutions, the functions calculating PSNR and
-  // SSIM return -1.0, instead of a positive value as usual. VideoAnalyzer
-  // aborts if the average psnr/ssim are below the given threshold, which is
-  // 0.0 by default. Setting the thresholds to -1.1 prevents the unnecessary
-  // abort.
-  VideoStream& selected_stream = params_.ss.streams[params_.ss.selected_stream];
+  std::set<uint32_t> excluded_ssrcs;
+  for (size_t i = 0; i < params_.ss.streams.size(); i++) {
+    if (i != params_.ss.selected_stream) {
+      excluded_ssrcs.insert(kVideoSendSsrcs[i]);
+      excluded_ssrcs.insert(kSendRtxSsrcs[i]);
+    }
+  }
+
+  test::LayerFilteringTransport send_transport(
+      params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
+      params_.video.selected_tl, params_.ss.selected_sl,
+      excluded_ssrcs);
+  test::DirectTransport recv_transport(params_.pipe, receiver_call_.get());
+
 
   bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
   VideoAnalyzer analyzer(
@@ -1584,7 +1589,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
   // calls.
   test::LayerFilteringTransport transport(
       params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
-      params.video.selected_tl, params_.ss.selected_sl);
+      params.video.selected_tl, params_.ss.selected_sl,
+      std::set<uint32_t>());
   // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
   // least share as much code as possible. That way this test would also match
   // the full stack tests better.
@@ -1617,8 +1623,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
     if (params_.audio.enabled && params_.audio.sync_video)
       video_receive_configs_[stream_id].sync_group = kSyncGroup;
 
-    if (params_.screenshare.enabled)
-      SetupScreenshareOrSVC();
+    SetupScreenshareOrSVC();
 
     video_send_stream_ = call->CreateVideoSendStream(
         video_send_config_.Copy(), video_encoder_config_.Copy());