Add muted_output parameter to ACM

webrtc/modules/audio_coding/acm2/acm_receiver.cc

Index: webrtc/modules/audio_coding/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 57d7bd344a1c7dd7ca3bf3ba36cfdc6f8e0d9313..8c07631e3a62d46f7589c80890076b2e6e6a11c4 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -132,16 +132,16 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
   return 0;
 }
 
-int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
+int AcmReceiver::GetAudio(int desired_freq_hz,
+                          AudioFrame* audio_frame,
+                          bool* muted) {
   // Accessing members, take the lock.
   rtc::CritScope lock(&crit_sect_);
 
-  bool muted;
-  if (neteq_->GetAudio(audio_frame, &muted) != NetEq::kOK) {
+  if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
-  RTC_DCHECK(!muted);
 
   const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();