Propagate muted parameter to VoE::Channel

webrtc/modules/audio_coding/test/insert_packet_with_timing.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 123 matching lines @@
     time_to_playout_audio_ms_--;
     sender_clock_->AdvanceTimeMilliseconds(1);
     receiver_clock_->AdvanceTimeMilliseconds(1);
 
     // Reset action.
     *action = 0;
 
     // Is it time to pull audio?
     if (time_to_playout_audio_ms_ == 0) {
       time_to_playout_audio_ms_ = kPlayoutPeriodMs;
+      bool muted;
       receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
-                                    &frame_);
+                                    &frame_, &muted);
+      ASSERT_FALSE(muted);
       fwrite(frame_.data_, sizeof(frame_.data_[0]),
              frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
       *action |= kAudioPlayedOut;
     }
 
     // Is it time to push in next packet?
     if (time_to_insert_packet_ms_ <= .5) {
       *action |= kPacketPushedIn;
 
       // Update time-to-insert packet.
@@ skipping 143 matching lines @@
       if (delay_log != NULL) {
         fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
       }
     }
   }
   std::cout << std::endl;
   test.TearDown();
   if (delay_log != NULL)
     fclose(delay_log);
 }