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..43a9575562c80718c0e77af942b81a9ab66c2cbe 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -20,9 +20,14 @@ namespace media {
 class AudioBus;
 }
 
+namespace WebKit {
+class WebAudioSourceProvider;
+}
+
 namespace content {
 
 class WebRtcLocalAudioRenderer;
+class WebRtcLocalAudioSourceProvider;
 class WebRtcLocalAudioTrack;
 
 // This class manages the capture data flow by getting data from its
@@ -73,6 +78,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
       media::ChannelLayout channel_layout,
       float sample_rate);
 
+  void EnablePeerConnectionMode();

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

documentation

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

+
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
@@ -95,6 +102,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   const std::string& device_id() const { return device_id_; }
 
+  WebKit::WebAudioSourceProvider* AudioSourceProvider() const;

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

GetAudioSourceProvider

usually for inline getters this would be:
audio_source_provider(), but since this isn't inline we use Get + camel caps

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

+
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   WebRtcAudioCapturer();
@@ -112,9 +121,9 @@ 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 +135,9 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // thread. It should NOT be called under |lock_|.
   void Stop();
 
+  // Helper function to get the buffer size based on |native_mode_| and

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

s/native_mode_/peer_connection_mode_

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

+  // 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 +152,16 @@ 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 values of utilized audio parameters.

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

nit: utilized and used does not mean exactly the same thing.  Suggest simply
"Cached audio parameters for output."

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

+  media::AudioParameters params_;
+
   bool running_;
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
+  int render_view_id_;
+
   // The media session ID used to identify which input device to be started by
   // the browser.
   int session_id_;
@@ -160,6 +173,15 @@ 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.
+  scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
+
+  // Flag which affects the buffer size used by the capturer.
+  bool peer_connection_mode_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };