Implements the GetSignalLevel and GetStats interface for the local audio track.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

 // Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 
 #include <vector>
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_vector.h"
 #include "base/synchronization/lock.h"
 #include "content/common/content_export.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }
 
 namespace webrtc {
 class AudioSourceInterface;
+class AudioProcessorInterface;
 }
 
 namespace content {
 
 class WebRtcAudioSinkAdapter;
 class WebRtcLocalAudioTrack;
 
 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
       NON_EXPORTED_BASE(
@@ skipping 10 matching lines @@
   virtual ~WebRtcLocalAudioTrackAdapter();
 
   void Initialize(WebRtcLocalAudioTrack* owner);
 
   std::vector<int> VoeChannels() const;
 
   // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
   // level of the audio data.
   void SetSignalLevel(int signal_level);
 
+  // Method called by the WebRtcLocalAudioTrack to set the processor that
+  // applies signal processing on the data of the track.
+  // This class will keep a reference of the |processor|.
+  // Called on the main render thread.
+  void SetAudioProcessor(
+      talk_base::scoped_refptr<webrtc::AudioProcessorInterface> processor);

[tommi (sloooow) - chröme, 2014/03/04 17:33:54]

do we need to use talk_base::scoped_refptr here?
I'd prefer to use chrome types wherever possible and reduce usage of third_party
types to only where we absolutely need it.

[no longer working on chromium, 2014/03/04 18:54:07]

Done.

Actually it is not a problem since adapter is designed to work with the
libjingle interface.

+
  private:
   // webrtc::MediaStreamTrack implementation.
   virtual std::string kind() const OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
   virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
+  virtual bool GetSignalLevel(int* level) OVERRIDE;
+  virtual talk_base::scoped_refptr<webrtc::AudioProcessorInterface>
+      GetAudioProcessor() OVERRIDE;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
   virtual void RemoveChannel(int channel_id) OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
   virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
   virtual cricket::AudioRenderer* GetRenderer() OVERRIDE;
 
   // Weak reference.
   WebRtcLocalAudioTrack* owner_;
 
   // The source of the audio track which handles the audio constraints.
   // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
   talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
+  // The audio processsor that applies audio processing on the data of audio
+  // track.
+  talk_base::scoped_refptr<webrtc::AudioProcessorInterface> audio_processor_;

[tommi (sloooow) - chröme, 2014/03/04 17:33:54]

scoped_refptr here as well

[no longer working on chromium, 2014/03/04 18:54:07]

Done.

+
   // A vector of WebRtc VoE channels that the capturer sends data to.
   std::vector<int> voe_channels_;
 
   // A vector of the peer connection sink adapters which receive the audio data
   // from the audio track.
   ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
 
-  // Protects |voe_channels_|.
+  // The amplitude of the signal.
+  int signal_level_;
+
+  // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
   mutable base::Lock lock_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_