Revert of Wire up send-side bandwidth estimation.

webrtc/video/screenshare_loopback.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.
  */
 
@@ skipping 121 matching lines @@
     "Name of a target graph data file. If set, no preview will be shown.");
 std::string OutputFilename() {
   return static_cast<std::string>(FLAGS_output_filename);
 }
 
 DEFINE_int32(duration, 60, "Duration of the test in seconds.");
 int DurationSecs() {
   return static_cast<int>(FLAGS_duration);
 }
 
-DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
-
 DEFINE_string(
     force_fieldtrials,
     "",
     "Field trials control experimental feature code which can be forced. "
     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
     " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
     "trials are separated by \"/\"");
 }  // namespace flags
 
 void Loopback() {
   FakeNetworkPipe::Config pipe_config;
   pipe_config.loss_percent = flags::LossPercent();
   pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
   pipe_config.queue_length_packets = flags::QueueSize();
   pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
   pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
 
   Call::Config::BitrateConfig call_bitrate_config;
   call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
   call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
   call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
 
   VideoQualityTest::Params params{
-      {flags::Width(), flags::Height(), flags::Fps(),
-       flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
-       flags::MaxBitrateKbps() * 1000, flags::Codec(),
-       flags::NumTemporalLayers(), flags::MinTransmitBitrateKbps() * 1000,
-       call_bitrate_config, flags::TLDiscardThreshold(),
-       flags::FLAGS_send_side_bwe},
+      {
+        flags::Width(),
+        flags::Height(),
+        flags::Fps(),
+        flags::MinBitrateKbps() * 1000,
+        flags::TargetBitrateKbps() * 1000,
+        flags::MaxBitrateKbps() * 1000,
+        flags::Codec(),
+        flags::NumTemporalLayers(),
+        flags::MinTransmitBitrateKbps() * 1000,
+        call_bitrate_config,
+        flags::TLDiscardThreshold()
+      },
       {},  // Video specific.
       {true, flags::SlideChangeInterval(), flags::ScrollDuration()},
       {"screenshare", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename()},
       pipe_config,
       flags::FLAGS_logs};
 
   VideoQualityTest test;
   if (flags::OutputFilename().empty())
     test.RunWithVideoRenderer(params);
   else
     test.RunWithAnalyzer(params);
 }
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
   google::ParseCommandLineFlags(&argc, &argv, true);
   webrtc::test::InitFieldTrialsFromString(
       webrtc::flags::FLAGS_force_fieldtrials);
   webrtc::test::RunTest(webrtc::Loopback);
   return 0;
 }