Switch LiveAudio to source provider solution.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index caa88d03825fe6d93ebf0f0c856c86405d9569d0..21246798cc4c6befeb04ed00ec81432abbdff57c 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -13,6 +13,7 @@
 #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"
 
@@ -50,6 +51,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   bool Initialize(int render_view_id,
                   media::ChannelLayout channel_layout,
                   int sample_rate,
+                  int buffer_size,
                   int session_id,
                   const std::string& device_id);
 
@@ -73,6 +75,11 @@ class CONTENT_EXPORT WebRtcAudioCapturer
       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);
@@ -95,6 +102,10 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   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();
@@ -112,9 +123,9 @@ class CONTENT_EXPORT WebRtcAudioCapturer
                        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
@@ -126,6 +137,9 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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_;
@@ -140,15 +154,20 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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_;
@@ -160,6 +179,15 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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);
 };