Wire up audio packet loss to BWE.

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 911 matching lines @@
     ss << ", " << params_.screenshare.scroll_duration << "s scroll";
   if (params_.ss.streams.size() > 1)
     ss << ", Stream #" << params_.ss.selected_stream;
   if (params_.ss.num_spatial_layers > 1)
     ss << ", Layer #" << params_.ss.selected_sl;
   ss << ")";
   return ss.str();
 }
 
 void VideoQualityTest::CheckParams() {
+  if (!params_.video.enabled)
+    return;
   // Add a default stream in none specified.
   if (params_.ss.streams.empty())
     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
@@ skipping 646 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