Adding stats that can be used to compute output audio levels.

webrtc/pc/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 @@
   rtc::scoped_refptr<MediaStreamTrackInterface> local_audio_track =
       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID",
                       MediaStreamTrackInterface::kEnded);
   local_stream->AddTrack(static_cast<AudioTrackInterface*>(
       local_audio_track.get()));
 
   cricket::VoiceSenderInfo voice_sender_info_ssrc1;
   voice_sender_info_ssrc1.local_stats.push_back(cricket::SsrcSenderInfo());
   voice_sender_info_ssrc1.local_stats[0].ssrc = 1;
   voice_sender_info_ssrc1.audio_level = 32767;
+  voice_sender_info_ssrc1.total_input_energy = 0.25;
+  voice_sender_info_ssrc1.total_input_duration = 0.5;
   voice_sender_info_ssrc1.echo_return_loss = 42;
   voice_sender_info_ssrc1.echo_return_loss_enhancement = 52;
 
   // Uses default values, the corresponding stats object should contain
   // undefined members.
   cricket::VoiceSenderInfo voice_sender_info_ssrc2;
   voice_sender_info_ssrc2.local_stats.push_back(cricket::SsrcSenderInfo());
   voice_sender_info_ssrc2.local_stats[0].ssrc = 2;
   voice_sender_info_ssrc2.audio_level = 0;
+  voice_sender_info_ssrc2.total_input_energy = 0.0;
+  voice_sender_info_ssrc2.total_input_duration = 0.0;
   voice_sender_info_ssrc2.echo_return_loss = -100;
   voice_sender_info_ssrc2.echo_return_loss_enhancement = -100;
 
   // Remote audio track
   rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio_track =
       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID",
                       MediaStreamTrackInterface::kLive);
   remote_stream->AddTrack(static_cast<AudioTrackInterface*>(
       remote_audio_track.get()));
 
   cricket::VoiceReceiverInfo voice_receiver_info;
   voice_receiver_info.local_stats.push_back(cricket::SsrcReceiverInfo());
   voice_receiver_info.local_stats[0].ssrc = 3;
   voice_receiver_info.audio_level = 16383;
+  voice_receiver_info.total_output_energy = 0.125;
+  voice_receiver_info.total_output_duration = 0.25;
 
   test_->CreateMockRtpSendersReceiversAndChannels(
       { std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1),
         std::make_pair(local_audio_track.get(), voice_sender_info_ssrc2) },
       { std::make_pair(remote_audio_track.get(), voice_receiver_info) },
       {}, {});
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCMediaStreamStats expected_local_stream(
@@ skipping 18 matching lines @@
                 RTCMediaStreamStats>());
 
   RTCMediaStreamTrackStats expected_local_audio_track_ssrc1(
       "RTCMediaStreamTrack_local_audio_LocalAudioTrackID_1",
       report->timestamp_us(), RTCMediaStreamTrackKind::kAudio);
   expected_local_audio_track_ssrc1.track_identifier = local_audio_track->id();
   expected_local_audio_track_ssrc1.remote_source = false;
   expected_local_audio_track_ssrc1.ended = true;
   expected_local_audio_track_ssrc1.detached = false;
   expected_local_audio_track_ssrc1.audio_level = 1.0;
+  expected_local_audio_track_ssrc1.total_audio_energy = 0.25;
+  expected_local_audio_track_ssrc1.total_samples_duration = 0.5;
   expected_local_audio_track_ssrc1.echo_return_loss = 42.0;
   expected_local_audio_track_ssrc1.echo_return_loss_enhancement = 52.0;
   ASSERT_TRUE(report->Get(expected_local_audio_track_ssrc1.id()));
   EXPECT_EQ(expected_local_audio_track_ssrc1,
             report->Get(expected_local_audio_track_ssrc1.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 
   RTCMediaStreamTrackStats expected_local_audio_track_ssrc2(
       "RTCMediaStreamTrack_local_audio_LocalAudioTrackID_2",
       report->timestamp_us(), RTCMediaStreamTrackKind::kAudio);
   expected_local_audio_track_ssrc2.track_identifier = local_audio_track->id();
   expected_local_audio_track_ssrc2.remote_source = false;
   expected_local_audio_track_ssrc2.ended = true;
   expected_local_audio_track_ssrc2.detached = false;
   expected_local_audio_track_ssrc2.audio_level = 0.0;
+  expected_local_audio_track_ssrc2.total_audio_energy = 0.0;
+  expected_local_audio_track_ssrc2.total_samples_duration = 0.0;
   // Should be undefined: |expected_local_audio_track_ssrc2.echo_return_loss|
   // and |expected_local_audio_track_ssrc2.echo_return_loss_enhancement|.
   ASSERT_TRUE(report->Get(expected_local_audio_track_ssrc2.id()));
   EXPECT_EQ(expected_local_audio_track_ssrc2,
             report->Get(expected_local_audio_track_ssrc2.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 
   RTCMediaStreamTrackStats expected_remote_audio_track(
       "RTCMediaStreamTrack_remote_audio_RemoteAudioTrackID_3",
       report->timestamp_us(), RTCMediaStreamTrackKind::kAudio);
   expected_remote_audio_track.track_identifier = remote_audio_track->id();
   expected_remote_audio_track.remote_source = true;
   expected_remote_audio_track.ended = false;
   expected_remote_audio_track.detached = false;
   expected_remote_audio_track.audio_level = 16383.0 / 32767.0;
+  expected_remote_audio_track.total_audio_energy = 0.125;
+  expected_remote_audio_track.total_samples_duration = 0.25;
   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 }
 
 TEST_F(RTCStatsCollectorTest,
        CollectRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video) {
   rtc::scoped_refptr<StreamCollection> local_streams =
       StreamCollection::Create();
@@ skipping 688 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc