Add basic support for "googDucking" to getUserMedia on Windows.

media/audio/win/audio_low_latency_input_win.h

 // Copyright (c) 2012 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.
 //
 // Implementation of AudioInputStream for Windows using Windows Core Audio
 // WASAPI for low latency capturing.
 //
 // Overview of operation:
 //
 // - An object of WASAPIAudioInputStream is created by the AudioManager
@@ skipping 70 matching lines @@
 class MEDIA_EXPORT WASAPIAudioInputStream
     : public AgcAudioStream<AudioInputStream>,
       public base::DelegateSimpleThread::Delegate,
       NON_EXPORTED_BASE(public base::NonThreadSafe) {
  public:
   // The ctor takes all the usual parameters, plus |manager| which is the
   // the audio manager who is creating this object.
   WASAPIAudioInputStream(AudioManagerWin* manager,
                          const AudioParameters& params,
                          const std::string& device_id);
+
   // The dtor is typically called by the AudioManager only and it is usually
   // triggered by calling AudioInputStream::Close().
   virtual ~WASAPIAudioInputStream();
 
   // Implementation of AudioInputStream.
   virtual bool Open() OVERRIDE;
   virtual void Start(AudioInputCallback* callback) OVERRIDE;
   virtual void Stop() OVERRIDE;
   virtual void Close() OVERRIDE;
   virtual double GetMaxVolume() OVERRIDE;
   virtual void SetVolume(double volume) OVERRIDE;
   virtual double GetVolume() OVERRIDE;
 
-  // Retrieves the sample rate used by the audio engine for its internal
-  // processing/mixing of shared-mode streams given a specifed device.
-  static int HardwareSampleRate(const std::string& device_id);
+  bool started() const { return started_; }
 
-  // Retrieves the number of audio channels used by the audio engine for its
-  // internal processing/mixing of shared-mode streams given a specified device.
-  static uint32 HardwareChannelCount(const std::string& device_id);
-
-  bool started() const { return started_; }
+  // Returns the default hardware audio parameters of the specific device.
+  static AudioParameters GetInputStreamParameters(const std::string& device_id);
 
  private:
   // DelegateSimpleThread::Delegate implementation.
   virtual void Run() OVERRIDE;
 
   // Issues the OnError() callback to the |sink_|.
   void HandleError(HRESULT err);
 
   // The Open() method is divided into these sub methods.
   HRESULT SetCaptureDevice();
@@ skipping 27 matching lines @@
   // is defined as the block of data which the user received in each
   // OnData() callback.
   size_t packet_size_frames_;
 
   // Size in bytes of each audio packet.
   size_t packet_size_bytes_;
 
   // Length of the audio endpoint buffer.
   uint32 endpoint_buffer_size_frames_;
 
+  // A copy of the supplied AudioParameter's |effects|.
+  const int effects_;
+
   // Contains the unique name of the selected endpoint device.
   // Note that AudioManagerBase::kDefaultDeviceId represents the default
   // device role and is not a valid ID as such.
   std::string device_id_;
 
   // Conversion factor used in delay-estimation calculations.
   // Converts a raw performance counter value to 100-nanosecond unit.
   double perf_count_to_100ns_units_;
 
   // Conversion factor used in delay-estimation calculations.
   // Converts from milliseconds to audio frames.
   double ms_to_frame_count_;
 
   // Pointer to the object that will receive the recorded audio samples.
   AudioInputCallback* sink_;
 
   // Windows Multimedia Device (MMDevice) API interfaces.
 
   // An IMMDevice interface which represents an audio endpoint device.
   base::win::ScopedComPtr<IMMDevice> endpoint_device_;
 
-  // Windows Audio Session API (WASAP) interfaces.
+  // Windows Audio Session API (WASAPI) interfaces.
 
   // An IAudioClient interface which enables a client to create and initialize
   // an audio stream between an audio application and the audio engine.
   base::win::ScopedComPtr<IAudioClient> audio_client_;
 
   // Loopback IAudioClient doesn't support event-driven mode, so a separate
   // IAudioClient is needed to receive notifications when data is available in
   // the buffer. For loopback input |audio_client_| is used to receive data,
   // while |audio_render_client_for_loopback_| is used to get notifications
   // when a new buffer is ready. See comment in InitializeAudioEngine() for
@@ skipping 16 matching lines @@
 
   // This event will be signaled when capturing shall stop.
   base::win::ScopedHandle stop_capture_event_;
 
   DISALLOW_COPY_AND_ASSIGN(WASAPIAudioInputStream);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_AUDIO_WIN_AUDIO_LOW_LATENCY_INPUT_WIN_H_