ACM/NetEq: Restructure how post-decode VAD is enabled

webrtc/modules/audio_coding/main/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 110 matching lines @@
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       id_(config.id),
       last_audio_decoder_(nullptr),
       previous_audio_activity_(AudioFrame::kVadPassive),
       current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
       audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       neteq_(NetEq::Create(config.neteq_config)),
-      vad_enabled_(true),
+      vad_enabled_(config.neteq_config.enable_post_decode_vad),
       clock_(config.clock),
       resampled_last_output_frame_(true),
       av_sync_(false),
       initial_delay_manager_(),
       missing_packets_sync_stream_(),
       late_packets_sync_stream_() {
   assert(clock_);
-
-  // Make sure we are on the same page as NetEq. Post-decode VAD is disabled by
-  // default in NetEq4, however, Audio Conference Mixer relies on VAD decision
-  // and fails if VAD decision is not provided.
-  if (vad_enabled_)
-    neteq_->EnableVad();
-  else
-    neteq_->DisableVad();
-
   memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
 }
 
 AcmReceiver::~AcmReceiver() {
   delete neteq_;
 }
 
 int AcmReceiver::SetMinimumDelay(int delay_ms) {
   if (neteq_->SetMinimumDelay(delay_ms))
@@ skipping 570 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc