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 173 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);

[the sun, 2016/09/16 17:18:36]

nit: DCHECK the pointer at function entry, since now you're dereffing it, not
just passing it on.

[hlundin-webrtc, 2016/09/20 07:48:37]

Done.

   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