Fix issue with zero rtt reports when using FlexFEC and add perf test.

webrtc/video/full_stack_tests.cc

 /*
  *  Copyright (c) 2013 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 118 matching lines @@
   foreman_cif.call.send_side_bwe = true;
   foreman_cif.video = {true, 352, 288, 30, 30000, 500000, 2000000, false, "VP8",
                        1, 0, 0, false, true, "", "foreman_cif"};
   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_flexfec", 0.0, 0.0,
                           kFullStackTestDurationSecs};
   foreman_cif.pipe.loss_percent = 5;
   foreman_cif.pipe.queue_delay_ms = 50;
   RunTest(foreman_cif);
 }
 
+TEST_F(FullStackTest, ForemanCif500kbpsPlr3Flexfec) {
+  VideoQualityTest::Params foreman_cif;
+  foreman_cif.call.send_side_bwe = true;
+  foreman_cif.video = {true,  352,          288, 30, 30000, 500000, 2000000,
+                       false, "VP8",        1,   0,  0,     false,  true,
+                       "",    "foreman_cif"};
+  foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_flexfec", 0.0,
+                          0.0, kFullStackTestDurationSecs};
+  foreman_cif.pipe.loss_percent = 3;
+  foreman_cif.pipe.link_capacity_kbps = 500;
+  foreman_cif.pipe.queue_delay_ms = 50;
+  RunTest(foreman_cif);
+}
+
+TEST_F(FullStackTest, ForemanCif500kbpsPlr3Ulpfec) {
+  VideoQualityTest::Params foreman_cif;
+  foreman_cif.call.send_side_bwe = true;
+  foreman_cif.video = {true,  352,          288, 30, 30000, 500000, 2000000,
+                       false, "VP8",        1,   0,  0,     true,   false,
+                       "",    "foreman_cif"};
+  foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_ulpfec", 0.0,
+                          0.0, kFullStackTestDurationSecs};
+  foreman_cif.pipe.loss_percent = 3;
+  foreman_cif.pipe.link_capacity_kbps = 500;
+  foreman_cif.pipe.queue_delay_ms = 50;
+  RunTest(foreman_cif);
+}
+
 #if defined(WEBRTC_USE_H264)
 TEST_F(FullStackTest, ForemanCifWithoutPacketlossH264) {
   // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
   VideoQualityTest::Params foreman_cif;
   foreman_cif.call.send_side_bwe = true;
   foreman_cif.video = {true, 352, 288, 30, 700000, 700000, 700000, false,
                        "H264", 1, 0, 0, false, false, "", "foreman_cif"};
   foreman_cif.analyzer = {"foreman_cif_net_delay_0_0_plr_0_H264", 0.0, 0.0,
                           kFullStackTestDurationSecs};
   RunTest(foreman_cif);
@@ skipping 578 matching lines @@
   std::vector<VideoStream> streams = {DefaultVideoStream(video_params_low),
                                       DefaultVideoStream(video_params_medium),
                                       DefaultVideoStream(video_params_high)};
   large_room.num_thumbnails = 50;
   large_room.ss = {streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
   RunTest(large_room);
 }
 
 
 }  // namespace webrtc