Add screenshare perf tests with lossy links

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <stdio.h>
@@ skipping 623 matching lines @@
     params_.ss.streams.push_back(VideoQualityTest::DefaultVideoStream(params_));
   if (params_.ss.num_spatial_layers == 0)
     params_.ss.num_spatial_layers = 1;
 
   if (params_.pipe.loss_percent != 0 ||
       params_.pipe.queue_length_packets != 0) {
     // Since LayerFilteringTransport changes the sequence numbers, we can't
     // use that feature with pack loss, since the NACK request would end up
     // retransmitting the wrong packets.
     RTC_CHECK(params_.ss.selected_sl == -1 ||
-              params_.ss.num_spatial_layers == 1);
+              params_.ss.selected_sl == params_.ss.num_spatial_layers - 1);
     RTC_CHECK(params_.common.selected_tl == -1 ||
-              params_.common.num_temporal_layers == 1);
+              params_.common.selected_tl ==
+                  params_.common.num_temporal_layers - 1);
   }
 
   // TODO(ivica): Should max_bitrate_bps == -1 represent inf max bitrate, as it
   // does in some parts of the code?
   RTC_CHECK_GE(params_.common.max_bitrate_bps,
                params_.common.target_bitrate_bps);
   RTC_CHECK_GE(params_.common.target_bitrate_bps,
                params_.common.min_bitrate_bps);
   RTC_CHECK_LT(params_.common.selected_tl, params_.common.num_temporal_layers);
   RTC_CHECK_LT(params_.ss.selected_stream, params_.ss.streams.size());
@@ skipping 407 matching lines @@
   send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc