Cleaned up the WebRtcAudioCapturer a bit.

content/renderer/media/webrtc_audio_capturer.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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
+#include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/tagged_list.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class MediaStreamAudioProcessor;
+class WebRtcAudioDeviceImpl;
 class WebRtcLocalAudioRenderer;
 class WebRtcLocalAudioTrack;
 
 // This class manages the capture data flow by getting data from its
 // |source_|, and passing it to its |tracks_|.
-// It allows clients to inject their own capture data source by calling
-// SetCapturerSource().
 // The threading model for this class is rather complex since it will be
 // created on the main render thread, captured data is provided on a dedicated
 // AudioInputDevice thread, and methods can be called either on the Libjingle
 // thread or on the main render thread but also other client threads
 // if an alternative AudioCapturerSource has been set.
 class CONTENT_EXPORT WebRtcAudioCapturer
     : public base::RefCountedThreadSafe<WebRtcAudioCapturer>,
       NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
-  // Use to construct the audio capturer.
+  // Used to construct the audio capturer. |render_view_id| specifies the
+  // render view consuming audio for capture, |render_view_id| as -1 is used
+  // by the unittests to skip creating a source via
+  // AudioDeviceFactory::NewInputDevice(), and allow injecting their own source
+  // via SetCapturerSourceForTesting() at a later state.  |device_info|
+  // contains all the device information that the capturer is created for.
+  // |constraints| contains the settings for audio processing.
+  // TODO(xians): Implement the interface for the audio source and move the
+  // |constraints| to ApplyConstraints().
   // Called on the main render thread.
-  static scoped_refptr<WebRtcAudioCapturer> CreateCapturer();
+  static scoped_refptr<WebRtcAudioCapturer> CreateCapturer(
+      int render_view_id,
+      const StreamDeviceInfo& device_info,
+      const blink::WebMediaConstraints& constraints,
+      WebRtcAudioDeviceImpl* audio_device);
 
-  // Creates and configures the default audio capturing source using the
-  // provided audio parameters.  |render_view_id| specifies the render view
-  // consuming audio for capture.  |session_id| is passed to the browser to
-  // decide which device to use.  |device_id| is used to identify which device
-  // the capturer is created for.  Called on the main render thread.
-  // TODO(xians): Implement the interface for the audio source and move the
-  // |constraints| to AddTrack().
-  bool Initialize(int render_view_id,
-                  media::ChannelLayout channel_layout,
-                  int sample_rate,
-                  int buffer_size,
-                  int session_id,
-                  const std::string& device_id,
-                  int paired_output_sample_rate,
-                  int paired_output_frames_per_buffer,
-                  int effects,
-                  const blink::WebMediaConstraints& constraints);
 
   // Add a audio track to the sinks of the capturer.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
   // other clients may call it from other threads. The current implementation
   // does not support multi-thread calling.
   // The first AddTrack will implicitly trigger the Start() of this object.
-  // Called on the main render thread or libjingle working thread.
-  // TODO(xians): Pass the track constraints via AddTrack().
   void AddTrack(WebRtcLocalAudioTrack* track);
 
   // Remove a audio track from the sinks of the capturer.
   // If the track has been added to the capturer, it  must call RemoveTrack()
   // before it goes away.
   // Called on the main render thread or libjingle working thread.
   void RemoveTrack(WebRtcLocalAudioTrack* track);
 
-  // SetCapturerSource() is called if the client on the source side desires to
-  // provide their own captured audio data. Client is responsible for calling
-  // Start() on its own source to have the ball rolling.
-  // Called on the main render thread.
-  void SetCapturerSource(
-      const scoped_refptr<media::AudioCapturerSource>& source,
-      media::ChannelLayout channel_layout,
-      float sample_rate,
-      int effects,
-      const blink::WebMediaConstraints& constraints);
-
   // Called when a stream is connecting to a peer connection. This will set
   // up the native buffer size for the stream in order to optimize the
   // performance for peer connection.
   void EnablePeerConnectionMode();
 
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
   int Volume() const;
   int MaxVolume() const;
 
-  bool is_recording() const { return running_; }
 
   // Audio parameters utilized by the source of the audio capturer.
   // TODO(phoglund): Think over the implications of this accessor and if we can
   // remove it.
   media::AudioParameters source_audio_parameters() const;
 
   // Gets information about the paired output device. Returns true if such a
   // device exists.
   bool GetPairedOutputParameters(int* session_id,
                                  int* output_sample_rate,
                                  int* output_frames_per_buffer) const;
 
-  const std::string& device_id() const { return device_id_; }
-  int session_id() const { return session_id_; }
+  const std::string& device_id() const { return device_info_.device.id; }
+  int session_id() const { return device_info_.session_id; }
 
   // Stops recording audio. This method will empty its track lists since
   // stopping the capturer will implicitly invalidate all its tracks.
   // This method is exposed to the public because the media stream track can
   // call Stop() on its source.
   void Stop();
 
   // Called by the WebAudioCapturerSource to get the audio processing params.
   // This function is triggered by provideInput() on the WebAudio audio thread,
   // TODO(xians): Remove after moving APM from WebRtc to Chrome.
   void GetAudioProcessingParams(base::TimeDelta* delay, int* volume,
                                 bool* key_pressed);
 
   // Called by the WebRtcAudioDeviceImpl to push the render audio to
   // audio processor for echo cancellation analysis.
   void FeedRenderDataToAudioProcessor(const int16* render_audio,
                                       int sample_rate,
                                       int number_of_channels,
                                       int number_of_frames,
                                       base::TimeDelta render_delay);
 
+  // Use by the unittests to inject their own source to the capturer.
+  void SetCapturerSourceForTesting(
+      const scoped_refptr<media::AudioCapturerSource>& source,
+      media::AudioParameters params);
+
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
-  WebRtcAudioCapturer();
   virtual ~WebRtcAudioCapturer();
 
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
 
+  WebRtcAudioCapturer(int render_view_id,
+                      const StreamDeviceInfo& device_info,
+                      const blink::WebMediaConstraints& constraints,
+                      WebRtcAudioDeviceImpl* audio_device);
+
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
                        double volume,
                        bool key_pressed) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
+  // Initializes the default audio capturing source using the provided render
+  // view id and device information. Return true if success, otherwise false.
+  bool Initialize();
+
+  // SetCapturerSource() is called if the client on the source side desires to
+  // provide their own captured audio data. Client is responsible for calling
+  // Start() on its own source to have the ball rolling.
+  // Called on the main render thread.
+  void SetCapturerSource(
+      const scoped_refptr<media::AudioCapturerSource>& source,
+      media::ChannelLayout channel_layout,
+      float sample_rate,
+      int effects,
+      const blink::WebMediaConstraints& constraints);
+
   // Starts recording audio.
   // Triggered by AddSink() on the main render thread or a Libjingle working
   // thread. It should NOT be called under |lock_|.
   void Start();
 
   // Helper function to get the buffer size based on |peer_connection_mode_|
   // and sample rate;
   int GetBufferSize(int sample_rate) const;
 
   // Used to DCHECK that we are called on the correct thread.
@@ skipping 15 matching lines @@
   blink::WebMediaConstraints constraints_;
 
   // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
   // data is in a unit of 10 ms data chunk.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   bool running_;
 
   int render_view_id_;
 
-  // Cached value for the hardware native buffer size, used when
-  // |peer_connection_mode_| is set to false.
-  int hardware_buffer_size_;
-
-  // The media session ID used to identify which input device to be started by
-  // the browser.
-  int session_id_;
-
-  // The device this capturer is given permission to use.
-  std::string device_id_;
+  // Cached information of the device used by the capturer.
+  const StreamDeviceInfo device_info_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   int volume_;
 
   // Flag which affects the buffer size used by the capturer.
   bool peer_connection_mode_;
 
-  int output_sample_rate_;
-  int output_frames_per_buffer_;
-
   // Cache value for the audio processing params.
   base::TimeDelta audio_delay_;
   bool key_pressed_;
 
   // Flag to help deciding if the data needs audio processing.
   bool need_audio_processing_;
 
+  // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
+  // of RenderThread.
+  WebRtcAudioDeviceImpl* audio_device_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_