Moved AudioUtil static functions to AudioManager interfaces

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 243cfd133c7ed32877e331799f09c45a6a49b302..ddb0ff6bc7862801267c164c0319c7d07346c2b7 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -51,7 +51,10 @@ static int GetBufferSizeForSampleRate(int sample_rate) {
   // output side.
   // TODO(henrika): It might be possible to reduce the input buffer
   // size and reduce the delay even more.
-  buffer_size = 2 * sample_rate / 100;
+  if (sample_rate == 44100)

[DaleCurtis, 2013/03/02 01:53:56]

Hmm, what is this?

[no longer working on chromium, 2013/03/04 14:55:04]

Oh, this change is making the webrtc audio work when the sample rate is 44100,
but it should be done in another CL.

Removed.

+    buffer_size = 2 * 440;
+  else
+    buffer_size = 2 * sample_rate / 100;
 #elif defined(OS_ANDROID)
   // TODO(leozwang): Tune and adjust buffer size on Android.
   buffer_size = 2 * sample_rate / 100;