Add new decoding statistics for muted output

webrtc/modules/audio_coding/acm2/call_statistics.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
-#include <assert.h>
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
-void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type) {
+void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type,
+                                    bool muted) {
   ++decoding_stat_.calls_to_neteq;
+  if (muted) {
+    ++decoding_stat_.decoded_muted_output;
+  }
   switch (speech_type) {
     case AudioFrame::kNormalSpeech: {
       ++decoding_stat_.decoded_normal;
       break;
     }
     case AudioFrame::kPLC: {
       ++decoding_stat_.decoded_plc;
       break;
     }
     case AudioFrame::kCNG: {
       ++decoding_stat_.decoded_cng;
       break;
     }
     case AudioFrame::kPLCCNG: {
       ++decoding_stat_.decoded_plc_cng;
       break;
     }
     case AudioFrame::kUndefined: {
       // If the audio is decoded by NetEq, |kUndefined| is not an option.
-      assert(false);
+      RTC_NOTREACHED();
     }
   }
 }
 
 void CallStatistics::DecodedBySilenceGenerator() {
   ++decoding_stat_.calls_to_silence_generator;
 }
 
 const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
   return decoding_stat_;
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc