Add muted_output parameter to ACM

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 114 matching lines @@
   if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
       0) {
     LOG(LERROR) << "AcmReceiver::InsertPacket "
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
     return -1;
   }
   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();
 
   // Update if resampling is required.
   const bool need_resampling =
       (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
 
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
@@ skipping 256 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc