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

webrtc/pc/rtcstats_integrationtest.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 534 matching lines @@
     verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.packets_received);
     verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.bytes_received);
     verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.packets_lost);
     if (inbound_stream.media_type.is_defined() &&
         *inbound_stream.media_type == "video") {
       verifier.TestMemberIsUndefined(inbound_stream.jitter);
     } else {
       verifier.TestMemberIsNonNegative<double>(inbound_stream.jitter);
     }
     verifier.TestMemberIsNonNegative<double>(inbound_stream.fraction_lost);
+    verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
     verifier.TestMemberIsUndefined(inbound_stream.packets_discarded);
     verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
     verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
     verifier.TestMemberIsUndefined(inbound_stream.burst_packets_discarded);
     verifier.TestMemberIsUndefined(inbound_stream.burst_loss_count);
     verifier.TestMemberIsUndefined(inbound_stream.burst_discard_count);
     verifier.TestMemberIsUndefined(inbound_stream.burst_loss_rate);
     verifier.TestMemberIsUndefined(inbound_stream.burst_discard_rate);
     verifier.TestMemberIsUndefined(inbound_stream.gap_loss_rate);
     verifier.TestMemberIsUndefined(inbound_stream.gap_discard_rate);
@@ skipping 12 matching lines @@
     VerifyRTCRTPStreamStats(outbound_stream, &verifier);
     if (outbound_stream.media_type.is_defined() &&
         *outbound_stream.media_type == "video") {
       verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.qp_sum);
     } else {
       verifier.TestMemberIsUndefined(outbound_stream.qp_sum);
     }
     verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.packets_sent);
     verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
     verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
-    // TODO(hbos): Defined in video but not audio case. Why? crbug.com/669877
-    verifier.MarkMemberTested(outbound_stream.round_trip_time, true);
     if (outbound_stream.media_type.is_defined() &&
         *outbound_stream.media_type == "video") {
       verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
     } else {
       verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
 
   bool VerifyRTCTransportStats(
@@ skipping 46 matching lines @@
   // Any pending stats requests should have completed in the act of destroying
   // the peer connection.
   EXPECT_TRUE(stats_obtainer->report());
 }
 #endif  // HAVE_SCTP
 #endif  // !defined(THREAD_SANITIZER)
 
 }  // namespace
 
 }  // namespace webrtc