Propagate muted parameter to VoE::Channel

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.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 813 matching lines @@
   // GetAudio always returns 10 ms, at the requested sample rate.
   if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "PlayoutData failed, RecOut Failed");
     return -1;
   }
   audio_frame->id_ = id_;
   return 0;
 }
 
-int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
-                                           AudioFrame* audio_frame) {
-  bool muted;
-  int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
-  RTC_DCHECK(!muted);
-  return ret;
-}
-
 /////////////////////////////////////////
 //   Statistics
 //
 
 // TODO(turajs) change the return value to void. Also change the corresponding
 // NetEq function.
 int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
   receiver_.GetNetworkStatistics(statistics);
   return 0;
 }
@@ skipping 123 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc