Add a virtual beamforming audio device on ChromeOS.

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/files/file.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/public/common/media_stream_request.h"
 #include "content/renderer/media/aec_dump_message_filter.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"
 
 namespace blink {
 class WebMediaConstraints;
 }
 
@@ skipping 21 matching lines @@
 // on the getUserMedia constraints, processes the data and outputs it in a unit
 // of 10 ms data chunk.
 class CONTENT_EXPORT MediaStreamAudioProcessor :
     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
     NON_EXPORTED_BASE(public AudioProcessorInterface),
     NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
  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 blink::WebMediaConstraints& constraints,
-                            int effects,
-                            WebRtcPlayoutDataSource* playout_data_source);
+  MediaStreamAudioProcessor(
+      const blink::WebMediaConstraints& constraints,
+      const MediaStreamDevice::AudioDeviceParameters& input_params,
+      WebRtcPlayoutDataSource* playout_data_source);
 
   // Called when the format of the capture data has changed.
   // Called on the main render thread. The caller is responsible for stopping
   // the capture thread before calling this method.
   // After this method, the capture thread will be changed to a new capture
   // thread.
   void OnCaptureFormatChanged(const media::AudioParameters& source_params);
 
   // Pushes capture data in |audio_source| to the internal FIFO. Each call to
   // this method should be followed by calls to ProcessAndConsumeData() while
@@ skipping 50 matching lines @@
                      int sample_rate,
                      int audio_delay_milliseconds) override;
   void OnPlayoutDataSourceChanged() override;
 
   // webrtc::AudioProcessorInterface implementation.
   // This method is called on the libjingle thread.
   void GetStats(AudioProcessorStats* stats) override;
 
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
-      const blink::WebMediaConstraints& constraints, int effects);
+      const blink::WebMediaConstraints& constraints,
+      const MediaStreamDevice::AudioDeviceParameters& input_params);
 
   // Helper to initialize the capture converter.
   void InitializeCaptureFifo(const media::AudioParameters& input_format);
 
   // Helper to initialize the render converter.
   void InitializeRenderFifoIfNeeded(int sample_rate,
                                     int number_of_channels,
                                     int frames_per_buffer);
 
   // Called by ProcessAndConsumeData().
@@ skipping 58 matching lines @@
   // Object for logging echo information when the AEC is enabled. Accessible by
   // the libjingle thread through GetStats().
   scoped_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_