fixed VP8 simulcast to not decode non-selected streams

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 "webrtc/video/video_quality_test.h"
@@ skipping 1407 matching lines @@
     RTC_CHECK(graph_data_output_file)
         << "Can't open the file " << params_.analyzer.graph_data_output_filename
         << "!";
   }
 
   webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log_);
   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(
       &send_transport, params_.analyzer.test_label,
 
       params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
       is_quick_test_enabled
           ? kFramesSentInQuickTest
           : params_.analyzer.test_durations_secs * params_.video.fps,
       graph_data_output_file, graph_title,
@@ skipping 111 matching lines @@
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
 
   std::unique_ptr<Call> call(Call::Create(call_config));
 
   // TODO(minyue): consider if this is a good transport even for audio only
   // 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.
   transport.SetReceiver(call->Receiver());
 
   VideoReceiveStream* video_receive_stream = nullptr;
   FlexfecReceiveStream* flexfec_receive_stream = nullptr;
   std::unique_ptr<test::VideoRenderer> local_preview;
   std::unique_ptr<test::VideoRenderer> loopback_video;
   if (params_.video.enabled) {
@@ skipping 12 matching lines @@
     loopback_video.reset(test::VideoRenderer::Create(
         title.c_str(), params_.ss.streams[stream_id].width,
         params_.ss.streams[stream_id].height));
 
     SetupVideo(&transport, &transport);
     video_send_config_.pre_encode_callback = local_preview.get();
     video_receive_configs_[stream_id].renderer = loopback_video.get();
     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());
     if (params_.video.flexfec) {
       RTC_DCHECK(!flexfec_receive_configs_.empty());
       flexfec_receive_stream =
           call->CreateFlexfecReceiveStream(flexfec_receive_configs_[0]);
     }
     video_receive_stream = call->CreateVideoReceiveStream(
         video_receive_configs_[stream_id].Copy());
@@ skipping 81 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc