Call OnSentPacket for all packets sent in the test framework.

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 229 matching lines @@
       printf("- Farewell, sweet Concorde!\n");
 
     // Signal stats polling thread if that is still waiting and stop it now,
     // since it uses the send_stream_ reference that might be reclaimed after
     // returning from this method.
     done_->Set();
     EXPECT_TRUE(stats_polling_thread_->Stop());
   }
 
   VideoCaptureInput* input_;
-  Transport* transport_;
+  Transport* const transport_;
   PacketReceiver* receiver_;
   VideoSendStream* send_stream_;
 
  private:
   struct FrameComparison {
     FrameComparison()
         : dropped(false),
           send_time_ms(0),
           recv_time_ms(0),
           render_time_ms(0),
@@ skipping 488 matching lines @@
 
   FILE* graph_data_output_file = nullptr;
   if (!params.analyzer.graph_data_output_filename.empty()) {
     graph_data_output_file =
         fopen(params.analyzer.graph_data_output_filename.c_str(), "w");
     RTC_CHECK(graph_data_output_file != nullptr)
         << "Can't open the file "
         << params.analyzer.graph_data_output_filename << "!";
   }
 
+  Call::Config call_config;
+  call_config.bitrate_config = params.common.call_bitrate_config;
+  CreateCalls(call_config, call_config);
+
   test::LayerFilteringTransport send_transport(
-      params.pipe, kPayloadTypeVP8, kPayloadTypeVP9,
+      params.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       static_cast<uint8_t>(params.common.tl_discard_threshold), 0);
-  test::DirectTransport recv_transport(params.pipe);
+  test::DirectTransport recv_transport(params.pipe, receiver_call_.get());
+
   VideoAnalyzer analyzer(
       &send_transport, params.analyzer.test_label,
       params.analyzer.avg_psnr_threshold, params.analyzer.avg_ssim_threshold,
       params.analyzer.test_durations_secs * params.common.fps,
       graph_data_output_file);
 
-  Call::Config call_config;
-  call_config.bitrate_config = params.common.call_bitrate_config;
-  CreateCalls(call_config, call_config);
-
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupFullStack(params, &analyzer, &recv_transport);
   send_config_.encoding_time_observer = &analyzer;
   receive_configs_[0].renderer = &analyzer;
   for (auto& config : receive_configs_)
     config.pre_decode_callback = &analyzer;
 
@@ skipping 37 matching lines @@
       test::VideoRenderer::Create("Loopback Video", params.common.width,
                                   params.common.height));
 
   // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
   // match the full stack tests.
   Call::Config call_config;
   call_config.bitrate_config = params.common.call_bitrate_config;
   rtc::scoped_ptr<Call> call(Call::Create(call_config));
 
   test::LayerFilteringTransport transport(
-      params.pipe, kPayloadTypeVP8, kPayloadTypeVP9,
+      params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       static_cast<uint8_t>(params.common.tl_discard_threshold), 0);
   // 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());
 
   SetupFullStack(params, &transport, &transport);
   send_config_.local_renderer = local_preview.get();
   receive_configs_[0].renderer = loopback_video.get();
 
@@ skipping 15 matching lines @@
   send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc