Add new decoding statistics for muted output

webrtc/modules/audio_coding/acm2/acm_receiver.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.
  */
 
@@ skipping 117 matching lines @@
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
     return -1;
   }
   return 0;
 }
 
 int AcmReceiver::GetAudio(int desired_freq_hz,
                           AudioFrame* audio_frame,
                           bool* muted) {
+  RTC_DCHECK(muted);
   // Accessing members, take the lock.
   rtc::CritScope lock(&crit_sect_);
 
   if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
 
   const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
 
@@ skipping 36 matching lines @@
   } else {
     resampled_last_output_frame_ = false;
     // We might end up here ONLY if codec is changed.
   }
 
   // Store current audio in |last_audio_buffer_| for next time.
   memcpy(last_audio_buffer_.get(), audio_frame->data_,
          sizeof(int16_t) * audio_frame->samples_per_channel_ *
              audio_frame->num_channels_);
 
-  call_stats_.DecodedByNetEq(audio_frame->speech_type_);
+  call_stats_.DecodedByNetEq(audio_frame->speech_type_, *muted);
   return 0;
 }
 
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
                               size_t channels,
                               int sample_rate_hz,
                               AudioDecoder* audio_decoder,
                               const std::string& name) {
   const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
@@ skipping 210 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc