WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

webrtc/modules/audio_coding/neteq/neteq_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 37 matching lines @@
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
 // longer required, this #define should be removed (and the code that it
 // enables).
 #define LEGACY_BITEXACT
 
 namespace webrtc {
 
-NetEqImpl::Dependencies::Dependencies(const NetEq::Config& config)
+NetEqImpl::Dependencies::Dependencies(
+    const NetEq::Config& config,
+    std::shared_ptr<AudioDecoderFactory> decoder_factory)
     : tick_timer(new TickTimer),
       buffer_level_filter(new BufferLevelFilter),
-      decoder_database(new DecoderDatabase(CreateBuiltinAudioDecoderFactory())),
+      decoder_database(new DecoderDatabase(std::move(decoder_factory))),
       delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
       delay_manager(new DelayManager(config.max_packets_in_buffer,
                                      delay_peak_detector.get(),
                                      tick_timer.get())),
       dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
       dtmf_tone_generator(new DtmfToneGenerator),
       packet_buffer(
           new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
       payload_splitter(new PayloadSplitter),
       timestamp_scaler(new TimestampScaler(*decoder_database)),
@@ skipping 2049 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc