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

webrtc/call/audio_send_stream.h

Index: webrtc/call/audio_send_stream.h
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 6c7e707d93bcd8b9710c7b20dc90fa7262be110a..57a2cdc82136807144737f506d790884c1f1768f 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -47,6 +47,9 @@ class AudioSendStream {
     int32_t jitter_ms = -1;
     int64_t rtt_ms = -1;
     int32_t audio_level = -1;
+    // See description of "totalAudioEnergy" in the WebRTC stats spec.
+    double total_input_energy = 0.0;
+    double total_input_duration = 0.0;
     float aec_quality_min = -1.0f;
     int32_t echo_delay_median_ms = -1;
     int32_t echo_delay_std_ms = -1;