WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

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 32 matching lines @@
                 *i == RentACodec::CodecId::kCNWB ||
                 *i == RentACodec::CodecId::kCNSWB ||
                 *i == RentACodec::CodecId::kCNFB));
 }
 
 }  // namespace
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : last_audio_decoder_(nullptr),
       last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
-      neteq_(NetEq::Create(config.neteq_config)),
+      neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
   assert(clock_);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
 }
 
 AcmReceiver::~AcmReceiver() {
   delete neteq_;
 }
 
@@ skipping 346 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc