Moves channel-dependent audio input processing to separate encoder task queue

webrtc/voice_engine/shared_data.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_SHARED_DATA_H
 #define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 class ProcessThread;
 
 namespace webrtc {
@@ skipping 11 matching lines @@
     ChannelManager& channel_manager() { return _channelManager; }
     AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); }
     void set_audio_device(
         const rtc::scoped_refptr<AudioDeviceModule>& audio_device);
     AudioProcessing* audio_processing() { return audioproc_.get(); }
     void set_audio_processing(AudioProcessing* audio_processing);
     TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
     OutputMixer* output_mixer() { return _outputMixerPtr; }
     rtc::CriticalSection* crit_sec() { return &_apiCritPtr; }
     ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
+    rtc::TaskQueue* encoder_queue();
 
     int NumOfSendingChannels();
     int NumOfPlayingChannels();
 
     // Convenience methods for calling statistics().SetLastError().
     void SetLastError(int32_t error) const;
     void SetLastError(int32_t error, TraceLevel level) const;
     void SetLastError(int32_t error, TraceLevel level,
                       const char* msg) const;
 
 protected:
-    const uint32_t _instanceId;
-    rtc::CriticalSection _apiCritPtr;
-    ChannelManager _channelManager;
-    Statistics _engineStatistics;
-    rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
-    OutputMixer* _outputMixerPtr;
-    TransmitMixer* _transmitMixerPtr;
-    std::unique_ptr<AudioProcessing> audioproc_;
-    std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
+ rtc::ThreadChecker construction_thread_;
+ const uint32_t _instanceId;
+ rtc::CriticalSection _apiCritPtr;
+ ChannelManager _channelManager;
+ Statistics _engineStatistics;
+ rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
+ OutputMixer* _outputMixerPtr;
+ TransmitMixer* _transmitMixerPtr;
+ std::unique_ptr<AudioProcessing> audioproc_;
+ std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
+ // |encoder_queue| is defined last to ensure all pending tasks are cancelled
+ // and deleted before any other members.
+ rtc::TaskQueue encoder_queue_ ACCESS_ON(construction_thread_);
 
-    SharedData();
-    virtual ~SharedData();
+ SharedData();
+ virtual ~SharedData();
 };
 
 }  // namespace voe
 }  // namespace webrtc
 #endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H