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

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 "webrtc/video/video_quality_test.h"
@@ skipping 1271 matching lines @@
 
   video_send_config_.rtp.extensions.clear();
   if (params_.call.send_side_bwe) {
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                      test::kTransportSequenceNumberExtensionId));
   } else {
     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;
 
   video_send_config_.suspend_below_min_bitrate =
       params_.video.suspend_below_min_bitrate;
 
   video_encoder_config_.number_of_streams = params_.ss.streams.size();
   video_encoder_config_.max_bitrate_bps = 0;
   for (size_t i = 0; i < params_.ss.streams.size(); ++i) {
     video_encoder_config_.max_bitrate_bps +=
         params_.ss.streams[i].max_bitrate_bps;
   }
   video_encoder_config_.video_stream_factory =
       new rtc::RefCountedObject<VideoStreamFactory>(params_.ss.streams);
 
   video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
   CreateMatchingReceiveConfigs(recv_transport);
 
   for (size_t i = 0; i < num_video_streams; ++i) {
     video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     video_receive_configs_[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
     video_receive_configs_[i].rtp.rtx_payload_types[payload_type] =
         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;
       decoder.decoder = new test::FakeDecoder();
       decoder.payload_type = video_send_config_.encoder_settings.payload_type;
       decoder.payload_name = video_send_config_.encoder_settings.payload_name;
       video_receive_configs_[i].decoders.clear();
       allocated_decoders_.emplace_back(decoder.decoder);
       video_receive_configs_[i].decoders.push_back(decoder);
     }
@@ skipping 555 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc