Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

webrtc/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-
-class AudioDecoder;
 class AudioSinkInterface;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioReceiveStream {
  public:
   struct Stats {
@@ skipping 57 matching lines @@
     // Identifier for an A/V synchronization group. Empty string to disable.
     // TODO(pbos): Synchronize streams in a sync group, not just one video
     // stream to one audio stream. Tracked by issue webrtc:4762.
     std::string sync_group;
 
     // Decoders for every payload that we can receive. Call owns the
     // AudioDecoder instances once the Config is submitted to
     // Call::CreateReceiveStream().
     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
     std::map<uint8_t, AudioDecoder*> decoder_map;
+
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   virtual Stats GetStats() const = 0;
 
   // Sets an audio sink that receives unmixed audio from the receive stream.
   // Ownership of the sink is passed to the stream and can be used by the
   // caller to do lifetime management (i.e. when the sink's dtor is called).
   // Only one sink can be set and passing a null sink clears an existing one.
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_