Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

webrtc/pc/rtcstatscollector.cc

 /*
  *  Copyright 2016 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.
  */
 
@@ skipping 243 matching lines @@
     const cricket::MediaSenderInfo& media_sender_info,
     RTCOutboundRTPStreamStats* outbound_stats) {
   RTC_DCHECK(outbound_stats);
   outbound_stats->ssrc = media_sender_info.ssrc();
   // TODO(hbos): Support the remote case. crbug.com/657856
   outbound_stats->is_remote = false;
   outbound_stats->packets_sent =
       static_cast<uint32_t>(media_sender_info.packets_sent);
   outbound_stats->bytes_sent =
       static_cast<uint64_t>(media_sender_info.bytes_sent);
-  if (media_sender_info.rtt_ms >= 0) {
-    outbound_stats->round_trip_time = static_cast<double>(
-        media_sender_info.rtt_ms) / rtc::kNumMillisecsPerSec;
-  }
 }
 
 void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
     const cricket::VoiceSenderInfo& voice_sender_info,
     RTCOutboundRTPStreamStats* outbound_audio) {
   SetOutboundRTPStreamStatsFromMediaSenderInfo(
       voice_sender_info, outbound_audio);
   outbound_audio->media_type = "audio";
   if (voice_sender_info.codec_payload_type) {
     outbound_audio->codec_id =
@@ skipping 969 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc