Chromium Code Reviews| 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. |
|
hbos
2017/07/10 09:58:24
Ditto link.
Zach Stein
2017/07/10 18:35:20
Done.
|
| + 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; |