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

webrtc/api/rtcstatscollector_unittest.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 1503 matching lines @@
       cricket::SsrcReceiverInfo());
   video_media_info.receivers[0].local_stats[0].ssrc = 1;
   video_media_info.receivers[0].packets_rcvd = 2;
   video_media_info.receivers[0].packets_lost = 42;
   video_media_info.receivers[0].bytes_rcvd = 3;
   video_media_info.receivers[0].fraction_lost = 4.5f;
   video_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42);
   video_media_info.receivers[0].firs_sent = 5;
   video_media_info.receivers[0].plis_sent = 6;
   video_media_info.receivers[0].nacks_sent = 7;
+  video_media_info.receivers[0].frames_decoded = 8;
 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
   codec_parameters.mime_type = "dummy";
   codec_parameters.clock_rate = 0;
   video_media_info.receive_codecs.insert(
       std::make_pair(codec_parameters.payload_type, codec_parameters));
 
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
@@ skipping 26 matching lines @@
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_InboundVideo_42";
   expected_video.fir_count = 5;
   expected_video.pli_count = 6;
   expected_video.nack_count = 7;
   expected_video.packets_received = 2;
   expected_video.bytes_received = 3;
   expected_video.packets_lost = 42;
   expected_video.fraction_lost = 4.5;
+  expected_video.frames_decoded = 8;
 
   ASSERT(report->Get(expected_video.id()));
   const RTCInboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*video.codec_id));
 }
 
@@ skipping 76 matching lines @@
   video_media_info.senders.push_back(cricket::VideoSenderInfo());
   video_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
   video_media_info.senders[0].local_stats[0].ssrc = 1;
   video_media_info.senders[0].firs_rcvd = 2;
   video_media_info.senders[0].plis_rcvd = 3;
   video_media_info.senders[0].nacks_rcvd = 4;
   video_media_info.senders[0].packets_sent = 5;
   video_media_info.senders[0].bytes_sent = 6;
   video_media_info.senders[0].rtt_ms = 7500;
   video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+  video_media_info.senders[0].frames_encoded = 8;
+  video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>(16);

[hta-webrtc, 2017/01/02 14:29:12]

How much work is it to add a test case for the case where qp_sum is not set?

[hbos, 2017/01/02 15:34:39]

Already covered :) Tested being set here in
CollectRTCOutboundRTPStreamStats_Video, tested not being set in
CollectRTCOutboundRTPStreamStats_Default below.

 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
   codec_parameters.mime_type = "dummy";
   codec_parameters.clock_rate = 0;
   video_media_info.send_codecs.insert(
       std::make_pair(codec_parameters.payload_type, codec_parameters));
 
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
@@ skipping 25 matching lines @@
   expected_video.media_type = "video";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
   expected_video.packets_sent = 5;
   expected_video.bytes_sent = 6;
   expected_video.round_trip_time = 7.5;
+  expected_video.frames_encoded = 8;
+  expected_video.qp_sum = 16;
 
   ASSERT(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*expected_video.codec_id));
 }
 
@@ skipping 21 matching lines @@
   video_media_info.senders.push_back(cricket::VideoSenderInfo());
   video_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
   video_media_info.senders[0].local_stats[0].ssrc = 1;
   video_media_info.senders[0].firs_rcvd = 2;
   video_media_info.senders[0].plis_rcvd = 3;
   video_media_info.senders[0].nacks_rcvd = 4;
   video_media_info.senders[0].packets_sent = 5;
   video_media_info.senders[0].bytes_sent = 6;
   video_media_info.senders[0].rtt_ms = -1;
   video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+  video_media_info.senders[0].frames_encoded = 7;
 
   EXPECT_CALL(*voice_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
 
   SessionStats session_stats;
   session_stats.proxy_to_transport["VoiceContentName"] = "TransportName";
   session_stats.proxy_to_transport["VideoContentName"] = "TransportName";
   session_stats.transport_stats["TransportName"].transport_name =
@@ skipping 39 matching lines @@
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
   expected_video.packets_sent = 5;
   expected_video.bytes_sent = 6;
+  expected_video.frames_encoded = 7;
   // |expected_video.round_trip_time| should be undefined.
+  // |expected_video.qp_sum| should be undefined.

[hbos, 2017/01/02 15:34:39]

^-- Here in CollectRTCOutboundRTPStreamStats_Default

 
   ASSERT(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
   std::unique_ptr<cricket::Candidate> rtp_local_candidate = CreateFakeCandidate(
       "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42);
@@ skipping 164 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc