Add histogram for end-to-end delay.

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 2035 matching lines @@
       }
       // RTX
       if (use_rtx_) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
         (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
             kSendRtxSsrcs[0];
         (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
             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 OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
@@ skipping 1435 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc