Moves ownership of OpenSL engine object to Android audio manager

webrtc/modules/audio_device/android/opensles_player.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.
  */
 
@@ skipping 37 matching lines @@
 // platform version 2.3) or later, then we can still use the OpenSL ES APIs but
 // the output latency may be higher.
 class OpenSLESPlayer {
  public:
   // The lower output latency path is used only if the application requests a
   // buffer count of 2 or more, and a buffer size and sample rate that are
   // compatible with the device's native output configuration provided via the
   // audio manager at construction.
   static const int kNumOfOpenSLESBuffers = 4;
 
-  // There is no need for this class to use JNI.
-  static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) {
-    return 0;
-  }
-  static void ClearAndroidAudioDeviceObjects() {}
-
   explicit OpenSLESPlayer(AudioManager* audio_manager);
   ~OpenSLESPlayer();
 
   int Init();
   int Terminate();
 
   int InitPlayout();
   bool PlayoutIsInitialized() const { return initialized_; }
 
   int StartPlayout();
@@ skipping 22 matching lines @@
 
   // Configures the SL_DATAFORMAT_PCM structure.
   SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
                                           int sample_rate,
                                           size_t bits_per_sample);
 
   // Allocate memory for audio buffers which will be used to render audio
   // via the SLAndroidSimpleBufferQueueItf interface.
   void AllocateDataBuffers();
 
-  // Creates/destroys the main engine object and the SLEngineItf interface.
-  bool CreateEngine();
-  void DestroyEngine();
+  // Gets the SL Engine Interface from the existing global Engine object.
+  // This interface exposes creation methods of all the OpenSL ES object types.
+  bool GetEngineInterface();

[magjed_webrtc, 2016/05/31 11:24:07]

I don't think calling this function 'GetEngineInterface' is appropriate when you
don't return any engine. I think CreateEngine or InitEngine is better.

[henrika_webrtc, 2016/05/31 12:04:03]

Good point. However, the OpenSL ES API is a bit messy it would be complicated to
return the actual interface and the interface is fact not created. The existing
engine object has a set if implicit interface (already created) and we simply
get (and stores) it here. I changed the name to Obtain instead. Hope that is OK.

 
   // Creates/destroys the output mix object.
   bool CreateMix();
   void DestroyMix();
 
   // Creates/destroys the audio player and the simple-buffer object.
   // Also creates the volume object.
   bool CreateAudioPlayer();
   void DestroyAudioPlayer();
 
   SLuint32 GetPlayState() const;
 
   // Ensures that methods are called from the same thread as this object is
   // created on.
   rtc::ThreadChecker thread_checker_;
 
   // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
   // non-application thread which is not attached to the Dalvik JVM.
   // Detached during construction of this object.
   rtc::ThreadChecker thread_checker_opensles_;
 
+  // Raw pointer to the audio manager injected at construction. Used to cache
+  // audio parameters and to access the global SL engine object needed by the
+  // GetEngineInterface() method.
+  AudioManager* audio_manager_;

[magjed_webrtc, 2016/05/31 11:24:07]

Can you comment on the lifetime of AudioManager? It looks like it's created in
AudioDeviceModuleImpl::CreatePlatformSpecificObjects().

[henrika_webrtc, 2016/05/31 12:04:03]

Done.

+
   // Contains audio parameters provided to this class at construction by the
   // AudioManager.
   const AudioParameters audio_parameters_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
   AudioDeviceBuffer* audio_device_buffer_;
 
   bool initialized_;
   bool playing_;
@@ skipping 22 matching lines @@
   // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 240
   // in each callback (one every 5ms). This class can then ask for 240 and the
   // FineAudioBuffer will ask WebRTC for new data only every second callback
   // and also cach non-utilized audio.
   std::unique_ptr<FineAudioBuffer> fine_buffer_;
 
   // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
   // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
   int buffer_index_;
 
-  // The engine object which provides the SLEngineItf interface.
-  // Created by the global Open SL ES constructor slCreateEngine().
-  webrtc::ScopedSLObjectItf engine_object_;
-
   // This interface exposes creation methods for all the OpenSL ES object types.
   // It is the OpenSL ES API entry point.
   SLEngineItf engine_;
 
   // Output mix object to be used by the player object.
   webrtc::ScopedSLObjectItf output_mix_;
 
   // The audio player media object plays out audio to the speakers. It also
   // supports volume control.
   webrtc::ScopedSLObjectItf player_object_;
@@ skipping 11 matching lines @@
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
 
   // Last time the OpenSL ES layer asked for audio data to play out.
   uint32_t last_play_time_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_