Switch LiveAudio to source provider solution.

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 "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class WebRtcLocalAudioRenderer;
@@ skipping 17 matching lines @@
   static scoped_refptr<WebRtcAudioCapturer> CreateCapturer();
 
   // 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.
   bool Initialize(int render_view_id,
                   media::ChannelLayout channel_layout,
                   int sample_rate,
+                  int buffer_size,
                   int session_id,
                   const std::string& device_id);
 
   // 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.
   // Called on the main render thread or libjingle working thread.
   void AddTrack(WebRtcLocalAudioTrack* track);
 
   // Remove a audio track from the sinks of the capturer.
   // 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);
 
+  // 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;
 
   // Enables or disables the WebRtc AGC control.
   // Called from a Libjingle working thread.
   void SetAutomaticGainControl(bool enable);
 
   bool is_recording() const { return running_; }
 
   // Audio parameters utilized by the audio capturer. Can be utilized by
   // a local renderer to set up a renderer using identical parameters as the
   // capturer.
   // TODO(phoglund): This accessor is inherently unsafe since the returned
   // parameters can become outdated at any time. Think over the implications
   // of this accessor and if we can remove it.
   media::AudioParameters audio_parameters() const;
 
   const std::string& device_id() const { return device_id_; }
 
+  WebKit::WebAudioSourceProvider* audio_source_provider() const {
+    return source_provider_.get();
+  }
+
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   WebRtcAudioCapturer();
   virtual ~WebRtcAudioCapturer();
 
  private:
   class TrackOwner;
   typedef std::list<scoped_refptr<TrackOwner> > TrackList;
 
   // 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;
 
-  // Reconfigures the capturer with a new buffer size and capture parameters.
-  // Must be called without holding the lock. Returns true on success.
-  bool Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
+  // Reconfigures the capturer with a new capture parameters.
+  // Must be called without holding the lock.
+  void Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
 
   // 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();
 
   // Stops recording audio.
   // Triggered by RemoveSink() on the main render thread or a Libjingle working
   // thread. It should NOT be called under |lock_|.
   void Stop();
 
+  // 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.
   base::ThreadChecker thread_checker_;
 
   // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|,
   // |params_|, |buffering_| and |agc_is_enabled_|.
   mutable base::Lock lock_;
 
   // A list of audio tracks that the audio data is fed to.
   TrackList tracks_;
 
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
-  // Buffers used for temporary storage during capture callbacks.
-  // Allocated during initialization.
-  class ConfiguredBuffer;
-  scoped_refptr<ConfiguredBuffer> buffer_;
+  // Cached audio parameters for output.
+  media::AudioParameters params_;
+
   bool running_;
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
+  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_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   int volume_;
 
+  // The source provider to feed the capture data to other clients like
+  // WebAudio.
+  // TODO(xians): Move the source provider to track once we don't need to feed
+  // delay, volume, key_pressed information to WebAudioCapturerSource.
+  const scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
+
+  // Flag which affects the buffer size used by the capturer.
+  bool peer_connection_mode_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_