MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_audio_device_impl.h

Index: content/renderer/media/webrtc_audio_device_impl.h
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index a3bbf6b8ee7f362b696f0911bc66509d78317cc7..8999fc84f70783d910a425ec4338d1fca39ee23e 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -18,7 +18,6 @@
 #include "base/memory/scoped_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_not_impl.h"
 #include "ipc/ipc_platform_file.h"
 #include "media/base/audio_capturer_source.h"
@@ -183,7 +182,7 @@
 
 namespace content {
 
-class WebRtcAudioCapturer;
+class ProcessedLocalAudioSource;
 class WebRtcAudioRenderer;
 
 // TODO(xians): Move the following two interfaces to webrtc so that
@@ -306,15 +305,11 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // 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.
   // 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(ProcessedLocalAudioSource* capturer);
+  void RemoveAudioCapturer(ProcessedLocalAudioSource* 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
@@ -331,7 +326,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   }
 
  private:
-  typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
+  typedef std::list<ProcessedLocalAudioSource*> CapturerList;
   typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
   class RenderBuffer;