Propagate muted parameter to VoE::Channel

webrtc/modules/audio_coding/test/APITest.cc

 /*
  *  Copyright (c) 2012 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 305 matching lines @@
   return static_cast<APITest*>(obj)->APIRunA();
 }
 
 bool APITest::APIThreadB(void* obj) {
   return static_cast<APITest*>(obj)->APIRunB();
 }
 
 bool APITest::PullAudioRunA() {
   _pullEventA->Wait(100);
   AudioFrame audioFrame;
-  if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0) {
+  bool muted;
+  if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame, &muted) < 0) {
     bool thereIsDecoder;
     {
       ReadLockScoped rl(_apiTestRWLock);
       thereIsDecoder = _thereIsDecoderA;
     }
     if (thereIsDecoder) {
       fprintf(stderr, "\n>>>>>>    cannot pull audio A       <<<<<<<< \n");
     }
   } else {
     if (_writeToFile) {
       _outFileA.Write10MsData(audioFrame);
     }
   }
   return true;
 }
 
 bool APITest::PullAudioRunB() {
   _pullEventB->Wait(100);
   AudioFrame audioFrame;
-  if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0) {
+  bool muted;
+  if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame, &muted) < 0) {
     bool thereIsDecoder;
     {
       ReadLockScoped rl(_apiTestRWLock);
       thereIsDecoder = _thereIsDecoderB;
     }
     if (thereIsDecoder) {
       fprintf(stderr, "\n>>>>>>    cannot pull audio B       <<<<<<<< \n");
       fprintf(stderr, "%d %d\n", _testNumA, _testNumB);
     }
   } else {
@@ skipping 739 matching lines @@
   CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
   myChannel->ResetStats();
   {
     WriteLockScoped wl(_apiTestRWLock);
     *thereIsEncoder = true;
   }
   Wait(500);
 }
 
 }  // namespace webrtc