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

webrtc/call/audio_receive_stream.h

Index: webrtc/call/audio_receive_stream.h
diff --git a/webrtc/call/audio_receive_stream.h b/webrtc/call/audio_receive_stream.h
index e3bdd452473666bc68bebd16116944b84176035f..ffb5ba64a004786ea1cf9e217466bc48c2c32a4f 100644
--- a/webrtc/call/audio_receive_stream.h
+++ b/webrtc/call/audio_receive_stream.h
@@ -49,6 +49,10 @@ class AudioReceiveStream {
     uint32_t jitter_buffer_preferred_ms = 0;
     uint32_t delay_estimate_ms = 0;
     int32_t audio_level = -1;
+    // See description of "totalAudioEnergy" in the WebRTC stats spec:
+    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+    double total_output_energy = 0.0;
+    double total_output_duration = 0.0;
     float expand_rate = 0.0f;
     float speech_expand_rate = 0.0f;
     float secondary_decoded_rate = 0.0f;