Add content type information to encoded images and corresponding rtp extension header

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 2497 matching lines @@
         (*receive_configs)[0].rtp.rtx_payload_types[kFakeVideoSendPayloadType] =
             kSendRtxPayloadType;
       }
       // RTT needed for RemoteNtpTimeEstimator for the receive stream.
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
       encoder_config->content_type =
           screenshare_ ? VideoEncoderConfig::ContentType::kScreen
                        : VideoEncoderConfig::ContentType::kRealtimeVideo;
     }
 
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
+      frame_generator_capturer->SetFakeContentType(
+          screenshare_ ? VideoContentType::kScreenshare
+                       : VideoContentType::kDefault);
+    }
+
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
     }
 
     rtc::CriticalSection crit_;
@@ skipping 69 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("WebRTC.Video.EndToEndDelayInMs"));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs"));
+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs"));
   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"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
@@ skipping 1686 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