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

content/renderer/media/media_stream_audio_processor.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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "third_party/webrtc/modules/interface/module_common_types.h"
 
 namespace blink {
 class WebMediaConstraints;
 }
 
 namespace media {
 class AudioBus;
 class AudioFifo;
 class AudioParameters;
 }  // namespace media
 
 namespace webrtc {
 class AudioFrame;
 class TypingDetection;
 }
 
 namespace content {
 
 class RTCMediaConstraints;
 
+using webrtc::AudioProcessorInterface;
+
 // This class owns an object of webrtc::AudioProcessing which contains signal
 // processing components like AGC, AEC and NS. It enables the components based
 // on the getUserMedia constraints, processes the data and outputs it in a unit
 // of 10 ms data chunk.
 class CONTENT_EXPORT MediaStreamAudioProcessor :
-    public base::RefCountedThreadSafe<MediaStreamAudioProcessor>,
-    NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink) {
+    NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
+    NON_EXPORTED_BASE(public AudioProcessorInterface) {
  public:
   // |playout_data_source| is used to register this class as a sink to the
   // WebRtc playout data for processing AEC. If clients do not enable AEC,
   // |playout_data_source| won't be used.
   MediaStreamAudioProcessor(const media::AudioParameters& source_params,
                             const blink::WebMediaConstraints& constraints,
                             int effects,
                             WebRtcPlayoutDataSource* playout_data_source);
 
   // Pushes capture data in |audio_source| to the internal FIFO.
@@ skipping 37 matching lines @@
   friend class MediaStreamAudioProcessorTest;
 
   class MediaStreamAudioConverter;
 
   // WebRtcPlayoutDataSource::Sink implementation.
   virtual void OnPlayoutData(media::AudioBus* audio_bus,
                              int sample_rate,
                              int audio_delay_milliseconds) OVERRIDE;
   virtual void OnPlayoutDataSourceChanged() OVERRIDE;
 
+  // webrtc::AudioProcessorInterface implementation.
+  // This method is called on the libjingle thread.
+  virtual void GetStats(AudioProcessorStats* stats) OVERRIDE;
+
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
       const blink::WebMediaConstraints& constraints, int effects);
 
   // Helper to initialize the capture converter.
   void InitializeCaptureConverter(const media::AudioParameters& source_params);
 
   // Helper to initialize the render converter.
   void InitializeRenderConverterIfNeeded(int sample_rate,
                                          int number_of_channels,
@@ skipping 46 matching lines @@
 
   // Used to DCHECK that PushRenderData() is called on the render audio thread.
   base::ThreadChecker render_thread_checker_;
 
   // Flag to enable the stereo channels mirroring.
   bool audio_mirroring_;
 
   // Used by the typing detection.
   scoped_ptr<webrtc::TypingDetection> typing_detector_;
 
-  // Result from the typing detection.
-  bool typing_detected_;
+  // This flag is used to show the result of typing detection.
+  // It can be accessed by the capture audio thread and by the libjingle thread
+  // which calls GetStats().
+  base::subtle::Atomic32 typing_detected_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_