Moved injection of AudioDecoderFactory into voe::Channel.

webrtc/modules/audio_coding/include/audio_coding_module.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/deprecation.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // forward declarations
 struct CodecInst;
@@ skipping 34 matching lines @@
   struct Config {
     Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
       // Post-decode VAD is disabled by default in NetEq, however, Audio
       // Conference Mixer relies on VAD decisions and fails without them.
       neteq_config.enable_post_decode_vad = true;
     }
 
     int id;
     NetEq::Config neteq_config;
     Clock* clock;
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
   };
 
   ///////////////////////////////////////////////////////////////////////////
   // Creation and destruction of a ACM.
   //
   // The second method is used for testing where a simulated clock can be
   // injected into ACM. ACM will take the ownership of the object clock and
   // delete it when destroyed.
   //
   static AudioCodingModule* Create(int id);
@@ skipping 725 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_