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

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 455 matching lines @@
         verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
       }
       verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
       verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
       verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
       // Audio-only members should be undefined
       verifier.TestMemberIsUndefined(media_stream_track.audio_level);
       verifier.TestMemberIsUndefined(media_stream_track.echo_return_loss);
       verifier.TestMemberIsUndefined(
           media_stream_track.echo_return_loss_enhancement);
+      verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
+      verifier.TestMemberIsUndefined(media_stream_track.total_samples_duration);
     } else {
       RTC_DCHECK_EQ(*media_stream_track.kind,
                     RTCMediaStreamTrackKind::kAudio);
       // Video-only members should be undefined
       verifier.TestMemberIsUndefined(media_stream_track.frame_width);
       verifier.TestMemberIsUndefined(media_stream_track.frame_height);
       verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
       verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
       verifier.TestMemberIsUndefined(media_stream_track.frames_received);
       verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
       verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
       verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
       verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
       verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
       // Audio-only members
       verifier.TestMemberIsNonNegative<double>(media_stream_track.audio_level);
+      // Remote audio tracks are expected to have audio energy/duration values
+      // that can be used to compute audio levels between getStats calls.
+      // TODO(deadbeef): Implement for sent audio tracks as well.
+      if (media_stream_track.remote_source.is_defined() &&
+          *media_stream_track.remote_source) {
+        verifier.TestMemberIsNonNegative<double>(
+            media_stream_track.total_audio_energy);
+        verifier.TestMemberIsNonNegative<double>(
+            media_stream_track.total_samples_duration);
+      } else {
+        verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
+        verifier.TestMemberIsUndefined(
+            media_stream_track.total_samples_duration);
+      }
       // TODO(hbos): |echo_return_loss| and |echo_return_loss_enhancement| are
       // flaky on msan bot (sometimes defined, sometimes undefined). Should the
       // test run until available or is there a way to have it always be
       // defined? crbug.com/627816
       verifier.MarkMemberTested(media_stream_track.echo_return_loss, true);
       verifier.MarkMemberTested(
           media_stream_track.echo_return_loss_enhancement, true);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }
@@ skipping 139 matching lines @@
   caller_ = nullptr;
   // Any pending stats requests should have completed in the act of destroying
   // the peer connection.
   EXPECT_TRUE(stats_obtainer->report());
 }
 #endif  // HAVE_SCTP
 
 }  // namespace
 
 }  // namespace webrtc