Add new decoding statistics for muted output

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  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 2593 matching lines @@
     rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
     rinfo.accelerate_rate = stats.accelerate_rate;
     rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
     rinfo.decoding_calls_to_silence_generator =
         stats.decoding_calls_to_silence_generator;
     rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
     rinfo.decoding_normal = stats.decoding_normal;
     rinfo.decoding_plc = stats.decoding_plc;
     rinfo.decoding_cng = stats.decoding_cng;
     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
+    rinfo.decoding_muted_output = stats.decoding_muted_output;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
     std::unique_ptr<webrtc::AudioSinkInterface> sink) {
@@ skipping 36 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE