Add new decoding statistics for muted output

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 602 matching lines @@
         speech_expand_rate(0),
         secondary_decoded_rate(0),
         accelerate_rate(0),
         preemptive_expand_rate(0),
         decoding_calls_to_silence_generator(0),
         decoding_calls_to_neteq(0),
         decoding_normal(0),
         decoding_plc(0),
         decoding_cng(0),
         decoding_plc_cng(0),
+        decoding_muted_output(0),
         capture_start_ntp_time_ms(-1) {}
 
   int ext_seqnum;
   int jitter_ms;
   int jitter_buffer_ms;
   int jitter_buffer_preferred_ms;
   int delay_estimate_ms;
   int audio_level;
   // fraction of synthesized audio inserted through expansion.
   float expand_rate;
   // fraction of synthesized speech inserted through expansion.
   float speech_expand_rate;
   // fraction of data out of secondary decoding, including FEC and RED.
   float secondary_decoded_rate;
   // Fraction of data removed through time compression.
   float accelerate_rate;
   // Fraction of data inserted through time stretching.
   float preemptive_expand_rate;
   int decoding_calls_to_silence_generator;
   int decoding_calls_to_neteq;
   int decoding_normal;
   int decoding_plc;
   int decoding_cng;
   int decoding_plc_cng;
+  int decoding_muted_output;
   // Estimated capture start time in NTP time in ms.
   int64_t capture_start_ntp_time_ms;
 };
 
 struct VideoSenderInfo : public MediaSenderInfo {
   VideoSenderInfo()
       : packets_cached(0),
         firs_rcvd(0),
         plis_rcvd(0),
         nacks_rcvd(0),
@@ skipping 473 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_