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

webrtc/video/video_quality_test.cc

Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 96a488298e859229bbe2a80556c9b1f332e2f266..0ca9ac0e91886337737f13802a0fd14535c9ad4d 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1287,6 +1287,8 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
   }
+  video_send_config_.rtp.extensions.push_back(RtpExtension(
+      RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId));
 
   video_encoder_config_.min_transmit_bitrate_bps =
       params_.video.min_transmit_bps;
@@ -1314,6 +1316,8 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
         kSendRtxPayloadType;
     video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe;
     video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe;
+    // Enable RTT calculation so NTP time estimator will work.
+    video_receive_configs_[i].rtp.rtcp_xr.receiver_reference_time_report = true;
     // Force fake decoders on non-selected simulcast streams.
     if (i != params_.ss.selected_stream) {
       VideoReceiveStream::Decoder decoder;
@@ -1541,6 +1545,8 @@ void VideoQualityTest::CreateCapturer() {
         new test::FrameGeneratorCapturer(clock_, std::move(frame_generator_),
                                          params_.video.fps);
     EXPECT_TRUE(frame_generator_capturer->Init());
+    frame_generator_capturer->SetFakeContentType(
+        VideoContentType::kScreenshare);
     video_capturer_.reset(frame_generator_capturer);
   } else {
     if (params_.video.clip_name.empty()) {