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

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 816dfd18fc987a3995396e4afa31839f69a3eac8..8073cfeb047a707346257158c7036e8617dcb835 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -614,6 +614,8 @@ struct VoiceSenderInfo : public MediaSenderInfo {
       : ext_seqnum(0),
         jitter_ms(0),
         audio_level(0),
+        total_input_energy(0.0),
+        total_input_duration(0.0),
         aec_quality_min(0.0),
         echo_delay_median_ms(0),
         echo_delay_std_ms(0),
@@ -626,6 +628,9 @@ struct VoiceSenderInfo : public MediaSenderInfo {
   int ext_seqnum;
   int jitter_ms;
   int audio_level;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec.

[hbos, 2017/07/10 09:58:24]

nit: Provide the w3c.github.io-link (not rawgit-link), e.g:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioen...

[Zach Stein, 2017/07/10 18:35:20]

Done.

+  float total_input_energy;
+  float total_input_duration;

[hbos, 2017/07/10 09:58:24]

This should be double. The reason the other is float is probably because the old
getStats code use float and this file was initially written for that, but the
new (rtcstats_objects.h) supports and uses double, and the metrics are reported
as double before they get here.

[Zach Stein, 2017/07/10 18:35:20]

I changed this to make it work with the changes in statscollector.cc, which pass
this value along to StatsReport::AddFloat. Doesn't look like StatsReport
supports doubles. Would you prefer I cast the values in stascollector.cc? This
seemed more consistent with the existing code to me.

[hbos, 2017/07/11 08:01:01]

Yes, cast it to float in statscollector.cc before AddFloat. That way other stats
consumers/rtcstatscollector.cc are not limited in precision due to limitations
of statscollector.cc.

[Zach Stein, 2017/07/11 19:50:30]

Done.

   float aec_quality_min;
   int echo_delay_median_ms;
   int echo_delay_std_ms;
@@ -644,6 +649,8 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
         jitter_buffer_preferred_ms(0),
         delay_estimate_ms(0),
         audio_level(0),
+        total_output_energy(0.0),
+        total_output_duration(0.0),
         expand_rate(0),
         speech_expand_rate(0),
         secondary_decoded_rate(0),
@@ -664,6 +671,9 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   int jitter_buffer_preferred_ms;
   int delay_estimate_ms;
   int audio_level;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec.

[hbos, 2017/07/10 09:58:24]

Ditto link.

[Zach Stein, 2017/07/10 18:35:20]

Done.

+  float total_output_energy;
+  float total_output_duration;

[hbos, 2017/07/10 09:58:24]

Ditto double.

[Zach Stein, 2017/07/11 19:50:30]

Done.

   // fraction of synthesized audio inserted through expansion.
   float expand_rate;
   // fraction of synthesized speech inserted through expansion.