MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <stdint.h>
 
+#include <list>
 #include <string>
 #include <vector>
 
 #include "base/compiler_specific.h"
 #include "base/files/file.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/threading/thread_checker.h"
@@ skipping 279 matching lines @@
   int32_t PlayoutDelay(uint16_t* delay_ms) const override;
   int32_t RecordingDelay(uint16_t* delay_ms) const override;
   int32_t RecordingSampleRate(uint32_t* sample_rate) const override;
   int32_t PlayoutSampleRate(uint32_t* sample_rate) const override;
 
  public:
   // Sets the |renderer_|, returns false if |renderer_| already exists.
   // Called on the main renderer thread.
   bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
 
-  // Adds/Removes the capturer to the ADM.
+  // Adds/Removes the |capturer| to the ADM.  Does NOT take ownership.
+  // Capturers must remain valid until RemoveAudioCapturer() is called.
   // TODO(xians): Remove these two methods once the ADM does not need to pass
   // hardware information up to WebRtc.
-  void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
-  void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
-
-  // Gets the default capturer, which is the last capturer in |capturers_|.
-  // The method can be called by both Libjingle thread and main render thread.
-  scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
+  void AddAudioCapturer(WebRtcAudioCapturer* capturer);
+  void RemoveAudioCapturer(WebRtcAudioCapturer* capturer);
 
   // Gets paired device information of the capture device for the audio
   // renderer. This is used to pass on a session id, sample rate and buffer
   // size to a webrtc audio renderer (either local or remote), so that audio
   // will be rendered to a matching output device.
   // Returns true if the capture device has a paired output device, otherwise
   // false. Note that if there are more than one open capture device the
   // function will not be able to pick an appropriate device and return false.
   bool GetAuthorizedDeviceInfoForAudioRenderer(
       int* session_id, int* output_sample_rate, int* output_buffer_size);
 
   const scoped_refptr<WebRtcAudioRenderer>& renderer() const {
     return renderer_;
   }
 
  private:
-  typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
+  typedef std::list<WebRtcAudioCapturer*> CapturerList;
   typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
   class RenderBuffer;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   ~WebRtcAudioDeviceImpl() override;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioOutputDevice worker thread.
   void RenderData(media::AudioBus* audio_bus,
@@ skipping 12 matching lines @@
   // Used to check methods that run on the main render thread.
   base::ThreadChecker main_thread_checker_;
   // Used to check methods that are called on libjingle's signaling thread.
   base::ThreadChecker signaling_thread_checker_;
   base::ThreadChecker worker_thread_checker_;
   base::ThreadChecker audio_renderer_thread_checker_;
 
   mutable int ref_count_;
 
   // List of captures which provides access to the native audio input layer
-  // in the browser process.
+  // in the browser process.  The last capturer in this list is considered the
+  // "default capturer" by the methods implementing the
+  // webrtc::AudioDeviceModule interface.
   CapturerList capturers_;
 
   // Provides access to the audio renderer in the browser process.
   scoped_refptr<WebRtcAudioRenderer> renderer_;
 
   // A list of raw pointer of WebRtcPlayoutDataSource::Sink objects which want
   // to get the playout data, the sink need to call RemovePlayoutSink()
   // before it goes away.
   PlayoutDataSinkList playout_sinks_;
 
@@ skipping 27 matching lines @@
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16_t> render_buffer_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_