Piggybacking simulcast id and ALR experiment id into video content type extension.

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 2790 matching lines @@
     Call* receiver_call_;
     int64_t start_runtime_ms_;
     int num_frames_received_ GUARDED_BY(&crit_);
   } test(use_rtx, use_red, screenshare);
 
   metrics::Reset();
   RunBaseTest(&test);
 
   std::string video_prefix =
       screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
-
+  // The content type extension is disabled in non screenshare test,
+  // therefore no slicing on simulcast id should be present.
+  std::string video_suffix = screenshare ? ".S0" : "";
   // Verify that stats have been updated once.
   EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
   EXPECT_EQ(1, metrics::NumSamples(
                    "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
 
@@ skipping 41 matching lines @@
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
 
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
 
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs"));
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs"));
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs"));
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs"));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
+                                   video_suffix));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
+                                   video_suffix));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
+                                   video_suffix));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
+                                   video_suffix));
 
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
@@ skipping 2125 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc