RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.

webrtc/api/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 449 matching lines @@
       verifier->TestMemberIsUndefined(stream.pli_count);
       verifier->TestMemberIsUndefined(stream.nack_count);
     }
     verifier->TestMemberIsUndefined(stream.sli_count);
   }
 
   bool VerifyRTCInboundRTPStreamStats(
       const RTCInboundRTPStreamStats& inbound_stream) {
     RTCStatsVerifier verifier(report_, &inbound_stream);
     VerifyRTCRTPStreamStats(inbound_stream, &verifier);
+    verifier.TestMemberIsUndefined(inbound_stream.qp_sum);
     verifier.TestMemberIsDefined(inbound_stream.packets_received);
     verifier.TestMemberIsDefined(inbound_stream.bytes_received);
     verifier.TestMemberIsDefined(inbound_stream.packets_lost);
     if (inbound_stream.media_type.is_defined() &&
         *inbound_stream.media_type == "video") {
       verifier.TestMemberIsUndefined(inbound_stream.jitter);
     } else {
       verifier.TestMemberIsDefined(inbound_stream.jitter);
     }
     verifier.TestMemberIsDefined(inbound_stream.fraction_lost);
     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);
+    if (inbound_stream.media_type.is_defined() &&
+        *inbound_stream.media_type == "video") {
+      verifier.TestMemberIsDefined(inbound_stream.frames_decoded);
+    } else {
+      verifier.TestMemberIsUndefined(inbound_stream.frames_decoded);
+    }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
 
   bool VerifyRTCOutboundRTPStreamStats(
       const RTCOutboundRTPStreamStats& outbound_stream) {
     RTCStatsVerifier verifier(report_, &outbound_stream);
     VerifyRTCRTPStreamStats(outbound_stream, &verifier);
+    if (outbound_stream.media_type.is_defined() &&
+        *outbound_stream.media_type == "video") {
+      verifier.TestMemberIsDefined(outbound_stream.qp_sum);
+    } else {
+      verifier.TestMemberIsUndefined(outbound_stream.qp_sum);
+    }
     verifier.TestMemberIsDefined(outbound_stream.packets_sent);
     verifier.TestMemberIsDefined(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(
       const RTCTransportStats& transport) {
     RTCStatsVerifier verifier(report_, &transport);
     verifier.TestMemberIsDefined(transport.bytes_sent);
     verifier.TestMemberIsDefined(transport.bytes_received);
     verifier.TestMemberIsOptionalIDReference(
         transport.rtcp_transport_stats_id, RTCTransportStats::kType);
@@ skipping 34 matching lines @@
   // This will destroy the peer connection.
   caller_ = nullptr;
   // Any pending stats requests should have completed in the act of destroying
   // the peer connection.
   EXPECT_TRUE(stats_obtainer->report());
 }
 
 }  // namespace
 
 }  // namespace webrtc