Revert of Storing raw audio sink for default audio track.

webrtc/voice_engine/channel.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_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
@@ skipping 159 matching lines @@
     int32_t SetEngineInformation(
         Statistics& engineStatistics,
         OutputMixer& outputMixer,
         TransmitMixer& transmitMixer,
         ProcessThread& moduleProcessThread,
         AudioDeviceModule& audioDeviceModule,
         VoiceEngineObserver* voiceEngineObserver,
         CriticalSectionWrapper* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
-    void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink);
+    void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
 
     // API methods
 
     // VoEBase
     int32_t StartPlayout();
     int32_t StopPlayout();
     int32_t StartSend();
     int32_t StopSend();
     int32_t StartReceiving();
     int32_t StopReceiving();
@@ skipping 297 matching lines @@
     RtcEventLog* const event_log_;
 
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
     rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
     rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
-    rtc::scoped_refptr<AudioSinkInterface> audio_sink_;
+    rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
     // Downsamples to the codec rate if necessary.
     PushResampler<int16_t> input_resampler_;
     FilePlayer* _inputFilePlayerPtr;
     FilePlayer* _outputFilePlayerPtr;
     FileRecorder* _outputFileRecorderPtr;
     int _inputFilePlayerId;
     int _outputFilePlayerId;
@@ skipping 79 matching lines @@
     PacketRouter* packet_router_ = nullptr;
     rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
     rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
     rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_