Add new decoding statistics for muted output

webrtc/api/call/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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 39 matching lines @@
     float speech_expand_rate = 0.0f;
     float secondary_decoded_rate = 0.0f;
     float accelerate_rate = 0.0f;
     float preemptive_expand_rate = 0.0f;
     int32_t decoding_calls_to_silence_generator = 0;
     int32_t decoding_calls_to_neteq = 0;
     int32_t decoding_normal = 0;
     int32_t decoding_plc = 0;
     int32_t decoding_cng = 0;
     int32_t decoding_plc_cng = 0;
+    int32_t decoding_muted_output = 0;
     int64_t capture_start_ntp_time_ms = 0;
   };
 
   struct Config {
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
@@ skipping 60 matching lines @@
   // Sets playback gain of the stream, applied when mixing, and thus after it
   // is potentially forwarded to any attached AudioSinkInterface implementation.
   virtual void SetGain(float gain) = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_