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..d3a3e763d80a9cc056d02e8627eaf8bc8d2eacce 100644
--- a/webrtc/call/audio_receive_stream.h
+++ b/webrtc/call/audio_receive_stream.h
@@ -49,6 +49,9 @@ 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.
+    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;